| 00:00.04 | ChrisInSydney | that reads even more confusing |
| 00:00.52 | ChrisInSydney | i should get back to what I was doing...how are you anyway ? |
| 00:02.45 | ChrisInSydney | rrittgarn: 5410 is a digital hanset aAFAIK, 5610 is a H.323 I think |
| 00:02.47 | p3nguin | I probably rate fair. |
| 00:03.35 | ChrisInSydney | Are we talking full blonde or just highlights ? |
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| 00:05.23 | ChrisInSydney | rrittgarn: I remember seeing a SIP to digital handset box around. Something like an Astribank |
| 00:05.57 | ChrisInSydney | gave you support for digital hansets from Avaya, NEC etc |
| 00:06.17 | ChrisInSydney | I cant remember where I saw it, it was a couple of years ago, and it wasnt cheap |
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| 00:19.39 | rrittgarn | Ok so end result... tell the client to purchase a real phone. Thanks for the input. |
| 00:19.51 | ChrisInSydney | yup |
| 00:20.47 | ChrisInSydney | I did the sums and it worked out to over AU$200 per handset at the time and I think that we were around $0.88c to a greenback |
| 00:22.06 | ChrisInSydney | rrittgarn: Good luck. Must fly |
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| 00:26.15 | Dlukz | sorry about that p3nguin |
| 00:26.40 | Dlukz | p3nguin: the problem is that I am running execif, and it's acting really strange |
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| 00:28.33 | Dlukz | exten => h,n,ExecIf($["${Intro}" = "1"]?Set|HOYN=Yes,Set|HOYN=No) |
| 00:29.16 | p3nguin | dlukz: ExecIf() in 1.4 does not support "command to run if false." |
| 00:29.37 | p3nguin | And it uses commas, not pipelines. |
| 00:29.52 | Dlukz | ok, well i still get a "Could not find application! (HOYN=Yes)" |
| 00:30.12 | Dlukz | show application execif shows this |
| 00:30.14 | Dlukz | Usage: ExecIF (<expr>|<app>|<data>) |
| 00:30.20 | Dlukz | soo... :-\ |
| 00:30.34 | snadge | so i finally got the balls to setup asterisk without a gui.. and sent a 3 page fax to the usa via sip :P |
| 00:30.34 | p3nguin | exten => h,n,ExecIf($["${Intro}" = "1"],Set,HOYN=Yes) |
| 00:30.55 | Dlukz | ok, let me give that a shot |
| 00:31.12 | snadge | for bonus points id like to set up a fax/voice switch.. i think it already does that, but i havn't tested if that works |
| 00:31.42 | snadge | presumably it needs to answer the call immediately, detect if its a fax tone, then continue ringing if it isnt? |
| 00:31.42 | Dlukz | well...thank you p3nguin |
| 00:32.23 | Dlukz | i've been banging my head against this for a while..cause voip-info shows |
| 00:32.23 | Dlukz | Version 1.4.24.1 |
| 00:32.24 | Dlukz | ExecIF (<expr>?<app>|<data>) |
| 00:32.28 | p3nguin | To get the behavior for "if false," you can use another Set or you can use Set and a nested IF(). |
| 00:32.29 | Dlukz | i'm on 1.4.27 |
| 00:32.49 | Dlukz | p3nguin i'm just going to have a != "1" after it |
| 00:33.00 | Dlukz | or = "2" |
| 00:34.27 | tzafrir | OT: any idea what's the default password for Yealink VP-2009 video phone (telnet, user: root) ? |
| 00:34.27 | Nugget | telnet is eeeeeeevil! |
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| 00:36.07 | p3nguin | dlukz: Set(HOYN=${IF($["${Intro}" = "1"]?Yes:No)}) |
| 00:36.16 | p3nguin | I think that's much nicer. |
| 00:36.34 | p3nguin | exten => h,n,Set(HOYN=${IF($["${Intro}" = "1"]?Yes:No)}) |
| 00:37.00 | p3nguin | It mimics a more modern ExecIf(). |
| 00:38.13 | p3nguin | Of course that can be used in place of a modern ExecIf just as easily. |
| 00:40.56 | p3nguin | snadge: If you are using SIP, there's faxdetect. |
| 00:41.34 | p3nguin | snadge: You can share a voice DID with fax if you use faxdetect, or you can use a dedicated fax DID and not worry with detection. |
| 00:42.26 | p3nguin | snadge: In your ITSP's peer entry, set faxdetect=yes. In your incoming context, create extension 'fax' as seen in my example. |
| 00:52.05 | mistermocha | hey! don't knock telnet! |
| 00:52.28 | mistermocha | Nugget: I've worked on some amazing stuff in telnet |
| 00:52.54 | talntid | like MUD's! |
| 00:53.09 | mistermocha | on a separate note… |
| 00:53.17 | mistermocha | I'm shopping around for a conference room phone |
| 00:53.24 | mistermocha | I'm very familiar with the polycom soundpoint phones |
| 00:53.41 | mistermocha | I want to know if anyone has used other conference room phones, and how well they perform |
| 00:54.28 | talntid | I have used a polycom conference room phone. worked well |
| 00:54.31 | talntid | soundstation |
| 00:54.49 | mistermocha | talntid: as have I… I'm looking at other non-polycom phones |
| 00:55.52 | talntid | ah, you had said you used soundpoint, not soundstation. that's why I gavw my opinion :-) |
| 00:56.01 | talntid | I have not used any others. |
| 00:56.14 | mistermocha | ah, I did say soundpoint |
| 00:56.15 | mistermocha | my bad |
| 00:56.32 | talntid | and I didn't feel soundpoint would make a good conference phone.. :-) |
| 00:56.40 | mistermocha | no it doesn't |
| 00:56.45 | talntid | hehe |
| 00:57.07 | talntid | afk |
| 00:58.09 | Micc | I've seen a few others that were much cheaper than the polycom and said the worked as well, but I have no personal experience with them. And I forget what the name was. |
| 00:59.05 | mistermocha | the other manufacturers I've found so far are Konfitel and ClearOne |
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| 01:49.52 | ChrisInSydney | mistermocha: set up an old clear one for soneone ages ago. It was good, but only support numeric usernames and passwords !!! |
| 01:50.03 | ChrisInSydney | not very secure :-( |
| 01:50.15 | ChrisInSydney | Snom meeting point is nice |
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| 02:04.03 | ducdmannxx | hi all |
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| 02:10.31 | ducdmannxx | hi all |
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| 03:54.06 | asteriskmonkey | does asterisk 10 have changes to realtime so you no longer have to make create contexts with switch statments? ie can they be queried on the fly now, or does empty context with switch statements still need to live in the flat files? |
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| 05:19.15 | snadge | asterisk 10? what? lol |
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| 05:38.46 | ChannelZ | OMG! and I was like, and she was like |
| 05:44.38 | p3nguin | heh |
| 05:49.52 | p3nguin | ~asterisk10 |
| 05:49.52 | infobot | Asterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ |
| 05:50.31 | p3nguin | is running asterisk 8. |
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| 06:34.06 | butchevans2001 | I have installed chan_sccp_b on asterisk 1.8.9.2. I am trying to get the sccpdevice and lines via mysql. When asterisk starts, mysql is no seeing a mysql query happening and consequently, no devices are loaded. Is there an easy way to test (from asterisk) why this is happening? |
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| 07:14.36 | ChrisInSydney | hi all. Quick question. Is there a release of the astersik-extra-sounds in g722 or similar that isnt quite as crap as a gsm file that has been upsampled ? |
| 07:15.06 | ChrisInSydney | the g722 core sounds are great, the extras are a little hit and miss..... |
| 07:15.18 | ChrisInSydney | ? |
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| 07:21.48 | schmidts | good morning |
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| 07:23.43 | v0lZy | hello |
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| 07:45.13 | joako | I perform a Dial(...) and it returns a 302 moved and Asterisk re-directs the call correctly. Now, is there any way to capture the Diversion: header and pass it along to the new leg of the call? |
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| 08:50.30 | molnarp | hi, yesterday we've talked about a VPMADT032 module in a TDM410P card, which gave the"Failed: Sent 0 != ff VPMADT032 Failed HI page test" message when inserting its kernel module with the debug=255 parameter |
| 08:51.29 | molnarp | two of you guys suggested that this might be caused by the external power being not plugged in (as I have only FXO modules) |
| 08:51.35 | jsjc | is there a way without using queues of ringing two extensions at the same time? lets say that john has a SIP phone and a mobile and want to have both ringing at same time just in case he is not in his office... |
| 08:51.51 | molnarp | now I tried plugging in the power, but nothing changed |
| 08:53.54 | WIMPy | molnarp: That was just an idea that was easy to eclude as a potential problem. |
| 08:54.03 | v0lZy | hi WIMPy |
| 08:54.08 | WIMPy | I guess you have to contact Digium support. |
| 08:54.18 | WIMPy | Hi v0lZy |
| 08:54.47 | v0lZy | im still struggling |
| 08:54.51 | v0lZy | im beginning to hate my ISP |
| 08:55.08 | molnarp | i did, they wouldn't help, because i use debian stable, and it has older versions of asterisk and dahdi |
| 08:56.03 | WIMPy | Then uninstall it and make is yourself. |
| 08:56.16 | molnarp | i thought i'm going to put that card into another PC for a test, install latest asterisk and dahdi from source, and in this case Digium would help |
| 08:56.35 | molnarp | or, it would turn out that this is a HW problem |
| 08:57.06 | WIMPy | Possibly |
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| 08:57.54 | WIMPy | I have once been told that there is a possibility that certain DSP boards might not work with certain cards. |
| 09:00.37 | joako | jsjc, Just use & in the Dial() statement, e.g Dial(SIP/john&SIP/12125551212@provider,20,r). Only thing is if the mobile phone voicemail answers... |
| 09:02.46 | molnarp | WIMPy: thanks anyway |
| 09:03.13 | jsjc | perfect joako thanks.. |
| 09:03.30 | joako | molnarp, Why don't you try AsteriskNOW CD disc? |
| 09:04.05 | molnarp | it contains the latest version, and it's supported by Digium? |
| 09:05.16 | WIMPy | jsjc, joako: That's what the PSTN had CDO for, but I have never actually seen that on offer. |
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| 09:07.59 | joako | Any idea why I can't do pattern matching aginst the callerid "number" Unavailable? Works: exten => 12125551212/Restricted,1,Macro(joakopager); Doesn't work: exten => 12125551212/Unavailable,1,Macro(joakopager) |
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| 09:13.33 | Yourname` | Hello. Does anyone have a 'strategy' of some kind for new employees, hot-desking & ex-employees? For example, phones on every desk (dont know what they will register as), but each employee gets their own extension so no matter where they're seated, they have their extension in use.. and then when they quit, their extension gets assigned to some other new employee? |
| 09:14.56 | WIMPy | Like a manual login on the phone? |
| 09:15.31 | Dein | wdoekes2 ! |
| 09:16.13 | WIMPy | wonders what the phone tries to offer under the name 'intercept'. |
| 09:16.23 | Dein | wdoekes2: great thanks, your patch is ok, problem is solved :) |
| 09:17.32 | Yourname` | WIMPy: http://www.757.org/~joat/wiki/index.php?n=Main.SimpleHot-desking |
| 09:17.36 | Yourname` | Something like that.. |
| 09:17.43 | Yourname` | ...i guess.. :S |
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| 09:17.58 | mosez | hi |
| 09:19.16 | WIMPy | Yourname`: Do you have a need to assign extensions to phones as well, instead of just users? |
| 09:19.23 | mosez | what kind of card is required to run asterisk on an isdn "anlagenanschluss" from the german telekom? |
| 09:19.42 | WIMPy | mosez: Any cheap one |
| 09:20.04 | mosez | works a isdn card with hfc-s chip? its not a single isdn line... |
| 09:20.18 | WIMPy | mosez: Yes |
| 09:20.24 | joako | My CDR shows ,"""Unavailable "" <Unavailable>", but these calls are not handled by exten => 12125551212/Unavailable,1,Macro(joakopager) |
| 09:20.25 | joako | . What am I overlookling? |
| 09:20.44 | WIMPy | mosez: They are available with 1, 4 or 8 Ports. |
| 09:20.58 | Yourname` | WIMPy: Not really. Basically, each employee has a PC with eyeBeam. When it got installed, each eyeBeam got a 101 - 110 kind of extension. But lately, employees are moving desks. Each employee has their own clients ofcourse, so they try to leave voicemails to their corresponding employee's " |
| 09:21.06 | Yourname` | WIMPy: Not really. Basically, each employee has a PC with eyeBeam. When it got installed, each eyeBeam got a 101 - 110 kind of extension. But lately, employees are moving desks. Each employee has their own clients ofcourse, so they try to leave voicemails to their corresponding employee's "101" style extension. |
| 09:21.15 | WIMPy | There are 2 port ones as well, but that's either two 1 portss on the same PCB or a 4 port with two missing. |
| 09:22.25 | WIMPy | Yourname`: So they're on soft phones? That makes it dead easy. Well, if they log in to the PC that is. |
| 09:22.43 | joako | Yourname`, Why don't you upgrade to Bria and implement roaming profiles? |
| 09:22.45 | Yourname` | WIMPy: Yup. They log into the PC. They come in the morning, and they are given a PC. |
| 09:23.23 | mosez | wimpy: can i talk to you in german in a query? |
| 09:23.57 | WIMPy | Yourname`: Ok, so it should just work all by itself, if the softphone configuration is saved under the users PC OS account. |
| 09:24.13 | WIMPy | mosez: Yes, or you could join #asterisk-de |
| 09:24.59 | Yourname` | WIMPy: Yeah. PC1 has eyeBeam as "101". A new employee is hired and I give him "201 extension". I'd like him to come to PC1 and log in as 201. How can I make that possible? |
| 09:26.27 | WIMPy | Yourname`: If the users have personal accounts on the PC, the configuration of the phone will follow them just like any other configuration. |
| 09:27.01 | Yourname` | WIMPy: Let's not confuse the PC accounts. Right now it's just eyeBeam registering to Asterisk as 101. |
| 09:27.18 | WIMPy | Yourname`: The PC accounts are the key. |
| 09:27.39 | Yourname` | WIMPy: They are auto-deployed generic Windows 7 administrator accounts.. |
| 09:27.59 | WIMPy | Fix that. |
| 09:28.25 | WIMPy | Seriousely. They do not have any personal configuration? How does that work? |
| 09:30.23 | Yourname` | WIMPy: Because they just need browsers to access our cloud based CRM and have eyeBeam to answer their calls. |
| 09:31.29 | joako | As I understand it EyeBeam and Xlite store their configuration at %UserProfile%\AppData\Local |
| 09:31.52 | joako | ... Bria stores it at %UserProfile%\AppData\Roaming |
| 09:32.04 | WIMPy | Yourname`: Well, either you create accounts on the PC or yu have to come up with something like what you linked. |
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| 09:35.43 | Eckrall | morning all |
| 09:36.25 | Eckrall | does anyone know if 1.8 has an issue with callerID when the privacy flag is on? |
| 09:36.58 | Eckrall | has been fighting trying to get callerID(name) set on a call but the extension only ever recieves 'unknown' |
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| 09:46.41 | schmidts | Eckrall i have seen a bug depending to this lately |
| 09:48.14 | Eckrall | schmidts: ahh really, I have been looking, the only one I found was when there was no callerid(num) set, but I tried to work around this by forcing one, makes no difference |
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| 09:55.44 | *** join/#asterisk jofry (~procube@port-212-202-54-40.dynamic.qsc.de) |
| 09:56.02 | jofry | hi folks |
| 10:06.50 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 10:07.12 | *** join/#asterisk nzkiwi1 (~michaelnz@ip210-48-102-28.nettrust.net.nz) |
| 10:07.34 | jofry | I need some help with Asterisk 1.4.21 on debian |
| 10:07.41 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 10:07.52 | WIMPy | jofry: No, you need to upgrade :-) |
| 10:08.20 | jofry | WIMPy: I can't upgrade i have to use this version |
| 10:09.32 | ChannelZ | so what's the question |
| 10:09.37 | nzkiwi1 | I have cracked the problem with yealink tls. |
| 10:10.13 | jofry | The problem is that no phone can register at port 5060. I have checkt with tcpdump at 5060 an i see the sip traffic but asterisk does nothing |
| 10:10.54 | schmidts | Eckrall i dont know if it belongs to your problem but look here https://issues.asterisk.org/jira/browse/ASTERISK-19417 |
| 10:10.59 | WIMPy | jofry: Does it listen on the IP you're sending to? |
| 10:11.20 | jofry | WIMPy: yes |
| 10:12.18 | jofry | schimdts: thx |
| 10:12.26 | Eckrall | jofry: I assume you have also tried another model / softphone as a test |
| 10:12.51 | Eckrall | schmidts: superstar! thats exactly what I have been failing to find |
| 10:13.28 | jofry | I tried with sipsak from 3 other networks - i got 404 time out |
| 10:13.58 | schmidts | jofry 404 is not time out, it normally means target not found |
| 10:13.58 | Eckrall | other networks? |
| 10:14.10 | kaldemar | jofry: are the registering clients behind a NAT? |
| 10:14.44 | jofry | kaldemar: yes |
| 10:14.53 | kaldemar | ~sipnat |
| 10:14.53 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 10:15.38 | jofry | Eckrall: The asterisk is hosted in a datacenter. The clienst phones will register from 3 LAN networks |
| 10:15.54 | kaldemar | jofry: you need to configure them to be with nat=yes in sip.conf. |
| 10:16.29 | kaldemar | jofry: if your asterisk is behind a nat, you must also configure externip, localhost and nat=yes under [general] in sip.conf. more info behind the above links. |
| 10:16.32 | jofry | i set nat=yes for every user in sip.conf |
| 10:17.55 | kaldemar | jofry: enable sip debug (command "sip debug" or "sip set debug on") in asterisk's CLI and pastebin a registration attempt. |
| 10:18.01 | jofry | kaldemar: The asterisk server is hosted in datacenter without NAT an listen on a fix IP number |
| 10:19.06 | jofry | kaldemar: Ok - a will give sip debug a try - just a moment .. |
| 10:19.46 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 10:22.47 | *** join/#asterisk jsjc (~Adium@161.Red-83-45-143.dynamicIP.rima-tde.net) |
| 10:26.56 | Eckrall | jofry: have you set the localnet and externip in the general section as well? |
| 10:27.12 | ChrisInSydney | hey all |
| 10:28.01 | jofry | kaldemar: if have set debuggion with "sip debug" in CLI. But after i try to register with sipsak i see _nothing_ at CLI. Only some traffic from my SIP provider |
| 10:28.18 | ChrisInSydney | Quick question. Is there a release of the astersik-extra-sounds in g722 or similar that isnt quite as crap as a gsm file that has been upsampled ? the g722 core sounds are great, the extras are a little hit and miss..... |
| 10:32.38 | Yourname` | WIMPy: Thanks! |
| 10:36.54 | kaldemar | jofry: then asterisk is not getting the messages or they are completely misformed. |
| 10:40.06 | jofry | kaldemar: It looks so. But i wonder about incomming calls. There are ok at the same interface and port and i can see the traffic in debug mode :-( |
| 10:40.42 | joako | ChrisInSydney, I see there's an asterisk-sounds-extra-en-ulaw... but I don't like Alison and use a Pat Fleet set of sounds, so I can't comment on the quality |
| 10:41.21 | ChrisInSydney | Cheers. They all seem to sound a bit rough, like an upsample |
| 10:41.21 | nzkiwi1 | I have call-limit = 1 to work with BLF, but this stops me from transferring calls. please help |
| 10:41.45 | nzkiwi1 | worked fine before I set call limit = 1 |
| 10:41.45 | ChrisInSydney | nzkiwi1: Hey bro |
| 10:41.50 | nzkiwi1 | hi chris |
| 10:42.06 | nzkiwi1 | yealink tls problem solved |
| 10:42.12 | ChrisInSydney | cool shat was it ? |
| 10:42.26 | ChrisInSydney | what was it |
| 10:42.33 | ChrisInSydney | typo |
| 10:42.37 | ChrisInSydney | bad yupo |
| 10:42.40 | ChrisInSydney | typo |
| 10:42.50 | nzkiwi1 | v61 f/w (current) is crap but reverting back doesn't fix it unless other steps are taken to fix the phone up |
| 10:43.17 | nzkiwi1 | p/m yr email and I will send you the fix if u want it |
| 10:43.25 | ChrisInSydney | cool |
| 10:44.21 | joako | nzkiwi1, I believe I have call-limit=100 and my BLF certainly works |
| 10:45.24 | joako | ChrisInSydney, I can send you a copy of my sounds, maybe they are better. You can hear a brief demo on +1.530.3036245 |
| 10:46.04 | ChrisInSydney | joako: Cool |
| 10:47.20 | nzkiwi1 | oh ok I thought it had to be set at 1 to work. I will try that |
| 10:48.26 | ChrisInSydney | joako: Im a tight arse, have you got a SIP URI ? |
| 10:48.47 | joako | ChrisInSydney, can I send you a private message with that? |
| 10:49.16 | ChrisInSydney | cool |
| 10:53.13 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 10:54.24 | *** join/#asterisk qakhan (~qakhan@182.185.140.122) |
| 10:54.31 | qakhan | hi all |
| 10:55.22 | qakhan | i setup my asterisk on public ip and i installed a sip phone on android phone |
| 10:55.55 | qakhan | my ext get register on phone, but i cant hear the voice of caller |
| 10:56.17 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 10:58.16 | ChrisInSydney | qakhan: did you set nat=yes on your android reg in sip.conf ? |
| 10:59.02 | qakhan | yes i setup net=yes in sip.conf |
| 10:59.14 | qakhan | caller can hear me but i cant |
| 11:03.14 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:05.23 | jofry | kaldemar: I have pastebin the sip.conf |
| 11:08.30 | ChrisInSydney | qakhan. canreinvite=no ? |
| 11:09.55 | qakhan | no enable |
| 11:11.00 | ChrisInSydney | change that and give it a go |
| 11:13.13 | jofry | kaledemar: http://pastebin.com/XPsfqTLf |
| 11:23.18 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:24.37 | *** join/#asterisk qakhan (~qakhan@182.185.140.122) |
| 11:27.23 | qakhan | i made 100 exts 1000 to 1100. i want only ext 1003 can only dial ext 1012, no one cannot dial ext 1012 except ext 1003 |
| 11:27.28 | qakhan | any help |
| 11:32.32 | joako | qakhan, That can be done in the dialplan. |
| 11:33.48 | *** join/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 11:36.13 | ijpalmer | Morning all, I'm using *1.8 configured with realtime queues, when I set periodic announce frequency, the time between the message being played doesn't change. The phones ring for 15 seconds, then go silent and the message gets played after 20 seconds then loops round again. I've changed the setting to 5, 20 and 30 seconds but nothing changes. Any ideas anyone |
| 11:36.35 | qakhan | joako can u tell me how? |
| 11:41.29 | joako | qakhan, I don't really have time right now to explain it step-by-step. But if you have gotten this far, look in sip.conf under [1003] look for context=SOMETHING then look for [SOMETHING] in extensions.conf, you need to make a new one.... |
| 12:01.07 | *** join/#asterisk bitwize (~bitwize@h87-96-213-2.dynamic.se.alltele.net) |
| 12:02.27 | wanawan | jofry: |
| 12:02.48 | jofry | wanawan: ja |
| 12:11.39 | *** join/#asterisk cuco (~cuco@82.166.216.90) |
| 12:11.48 | cuco | tzafrir: ping |
| 12:12.17 | tzafrir | cuco, pong |
| 12:12.28 | cuco | tzafrir: wazzzzzzzzzzzaaaaaaaaaaaaa...p |
| 12:13.00 | tzafrir | 10 years old |
| 12:13.10 | tzafrir | :-) |
| 12:13.12 | cuco | shut up |
| 12:16.00 | *** join/#asterisk DaneoShiga (~dshiga@kraz.dreamhost.com) |
| 12:17.12 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 12:31.37 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
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| 12:35.46 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 12:40.02 | *** join/#asterisk kchehab (~kchehab@77.42.241.66) |
| 12:40.02 | kaldemar | jofry: comment out the deny/permit lines and try again. |
| 12:40.03 | *** join/#asterisk moy (~moy@216.172.42.74) |
| 12:40.05 | kchehab | hi |
| 12:40.14 | kchehab | i enable the recording to my extension and have no problem with that but the problem is that after a while the (.wav)files been deleted from /var/spool/asterisk/monitor/ in other hand the records still in the mysql and sure when you click download it will reply by no files found ,how to fix it ? |
| 12:40.15 | kchehab | regards |
| 12:41.32 | jofry | kaldermar: OK .. |
| 12:42.36 | kaldemar | jofry: and keep looking at asterisk CLI with sip debug enabled. |
| 12:50.45 | DaneoShiga | kchehab: maybe you should find what is deleting the recordings, maybe sothing in cron? then or you disable it, or you delete from the database as well |
| 12:55.49 | *** part/#asterisk bitwize (~bitwize@h87-96-213-2.dynamic.se.alltele.net) |
| 12:55.49 | *** join/#asterisk bitwize (~bitwize@h87-96-213-2.dynamic.se.alltele.net) |
| 13:00.05 | *** join/#asterisk z4nD4R (~zandar@pc-hqes208mn8citbe6mn3nptkoys67due.usr.iklub.sk) |
| 13:00.42 | z4nD4R | hi all :) |
| 13:01.19 | z4nD4R | i would ask you, if is possible that user can be "member" of 2 context (or more) |
| 13:02.02 | *** part/#asterisk Montys (~dmartinez@nat/digium/x-cfdqnctseidgaruq) |
| 13:02.04 | WIMPy | Use includes |
| 13:03.02 | z4nD4R | WIMPy: in extensions.conf? .. |
| 13:03.43 | *** join/#asterisk RickCogley (~anonymous@ntkngw522154.kngw.nt.ftth.ppp.infoweb.ne.jp) |
| 13:03.52 | z4nD4R | WIMPy: u mean this? https://wiki.asterisk.org/wiki/display/AST/Include+Statements+Basics ? |
| 13:04.35 | WIMPy | yes |
| 13:06.24 | z4nD4R | WIMPy: but this include all context, i need.. that only one user can be in 2 context... anyone (except that one user) can call users from another user....in other words, i need 2 separe group... but one user can calls to both |
| 13:06.42 | z4nD4R | *cant |
| 13:07.04 | WIMPy | So you make a third context that includes the other two. |
| 13:08.28 | z4nD4R | sorry but i dont understant how this can help me.. |
| 13:10.27 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:11.34 | z4nD4R | WIMPy: third context for user with can access to both groups? |
| 13:11.50 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 13:12.08 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
| 13:17.52 | qakhan | all i want to setup asterisk on public IP, but i dont want to use 5060 port on public ip. i want to use some other port like 5062 etc. but 5060 should be use internal network |
| 13:17.57 | qakhan | any help |
| 13:19.03 | kchehab | Deeewayne i check all cron jobs and nothing related to this issue ,btw i am using elastix |
| 13:19.27 | [TK]D-Fender | qakhan, set your bindport for 5062 in [general] and port=5060 in your peers. |
| 13:20.27 | kchehab | any idea how to monitor the script that is deleting these files ? |
| 13:20.39 | kchehab | Deeewayne ? |
| 13:21.00 | Deeewayne | kchehab, you were talking to DaneoShiga |
| 13:21.13 | qakhan | [TK]D-Fender i want only one ext to be register over public ip |
| 13:22.42 | DaneoShiga | kchehab: thinking... |
| 13:22.53 | [TK]D-Fender | qakhan, Actually... if you're expecting the inside phones to register TO 5060 on the server and outside on 5062 then this isn't going to work. You'll need to set up a proxy server between * and the outside to switch it to 5062 |
| 13:22.57 | z4nD4R | WIMPy: thx :) |
| 13:23.55 | *** join/#asterisk ccesario (~ccesario@187.17.166.162) |
| 13:24.01 | ccesario | #2600 |
| 13:24.23 | kchehab | DaneoShiga np waiting for your ideas |
| 13:25.01 | qakhan | [TK]D-Fender there no option bindport in sip.conf |
| 13:25.57 | [TK]D-Fender | qakhan, go read the sample. But either way it doesn't matter... you need * to actually bind to different port which is not happening.on it's own. You'll need to set up a proxy server to the outside to do what you're looking for. |
| 13:26.12 | [TK]D-Fender | qakhan, * won't do hat you need by itself |
| 13:26.15 | [TK]D-Fender | what* |
| 13:26.52 | qakhan | u meant should i add bindport=5062 in sip.conf? |
| 13:27.59 | [TK]D-Fender | qakhan, You aren't listening. I took that back. it won't do the job for you. |
| 13:28.27 | [TK]D-Fender | qakhan, You need a proxy server between * and the outside. |
| 13:29.06 | qakhan | listen you said "set your bindport for 5062 in [general] and port=5060 in your peers." |
| 13:29.16 | [TK]D-Fender | qakhan, I TOOK IT BACK |
| 13:29.23 | qakhan | no i am asking you "should i add bindport=5062 in sip.conf?" |
| 13:29.26 | DaneoShiga | kchehab: i think that maybe it's configurable throw the interface... google points to somehwere around ACD->Recordings |
| 13:29.28 | [TK]D-Fender | NO |
| 13:29.52 | *** part/#asterisk z4nD4R (~zandar@pc-hqes208mn8citbe6mn3nptkoys67due.usr.iklub.sk) |
| 13:30.02 | [TK]D-Fender | qakhan, FORGET what I said about bindport. You can't effective bind to MULTIPLE ports and what I first said is not applicable. |
| 13:30.17 | qakhan | ok |
| 13:30.35 | *** join/#asterisk clintc (~clintc@n128-227-215-193.xlate.ufl.edu) |
| 13:30.40 | *** part/#asterisk clintc (~clintc@n128-227-215-193.xlate.ufl.edu) |
| 13:30.43 | qakhan | so what i do now |
| 13:31.57 | [TK]D-Fender | qakhan, I have just told you 3 times |
| 13:33.01 | qakhan | ok a proxy server between * and outside network |
| 13:34.46 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 13:35.25 | qakhan | [TK]D-Fender can u help me how to setup proxy server |
| 13:35.39 | *** join/#asterisk manitou_ (~manitou@93.170.30.243) |
| 13:35.47 | [TK]D-Fender | qakhan, No. Go pick one and use their support services if you need help with it. |
| 13:36.21 | qakhan | ok |
| 13:42.07 | manitou_ | Hi guys. Has anyone experience in Asterisk for ARM Emdebian system compiling? |
| 13:42.21 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
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| 14:05.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 14:07.12 | kchehab | DaneoShiga can you please send me the url to see it |
| 14:10.33 | DaneoShiga | kchehab: nevermind, just noticed i've found that for trixbox |
| 14:19.20 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-bgrqjbxvxmhixeqb) |
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| 15:08.27 | *** join/#asterisk davidduffy (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 15:09.19 | davidduffy | hi all |
| 15:12.57 | _Corey_ | davidduffy: Is this David from the UK? |
| 15:12.58 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:13.00 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 15:13.42 | davidduffy | no sorry |
| 15:14.39 | _Corey_ | ah, welcome anyhow |
| 15:14.51 | davidduffy | thanks |
| 15:15.04 | davidduffy | though I live in the UK |
| 15:16.10 | *** join/#asterisk stephendowed (~stephen@197.255.215.252) |
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| 15:42.17 | stephendowed | cant asterisk do automatic transfer like is done on regular phone system like someone calls ma phone and i transfer it to another no or is the parking thing the only way? |
| 15:43.52 | [TK]D-Fender | stephendowed, Parking is a choice. Please clarify your intent in using the term "automatic transfer" |
| 15:43.58 | beek | stephendowed: you can transfer calls |
| 15:44.02 | beek | mornin' [TK]D-Fender |
| 15:44.12 | [TK]D-Fender | stephendowed, what is "automatic" about it? |
| 15:44.39 | [TK]D-Fender | stephendowed, The call spontaneously decides "nope ... this conversation isn't going anywhere" and tosses itself out? |
| 15:44.50 | stephendowed | okay lemme correct maself. i get a call and i want to just hit the transfer button on ma fone and it transfers to another extension |
| 15:45.01 | [TK]D-Fender | stYes, of course * supports this |
| 15:45.08 | [TK]D-Fender | stephendowed, Yes, of course * supports this |
| 15:45.21 | [TK]D-Fender | stephendowed, Thre are both ATTENDED transfers, and BLIND transfers |
| 15:45.26 | stephendowed | how do u achieve that in a dialplan if u dont mind telling a newbie |
| 15:45.37 | [TK]D-Fender | stephendowed, your PHONE does this |
| 15:45.40 | stephendowed | yes i know about attended and blind transfer |
| 15:45.57 | stephendowed | @fender so nothing to do on asterisk the fone just knows what to do? wow |
| 15:46.17 | stephendowed | just new to asterisk it seems everything needs to be achieved via d dialplan |
| 15:46.24 | [TK]D-Fender | stephendowed, phone says "toss this call to another URI I could dial" |
| 15:47.26 | [TK]D-Fender | stephendowed, What are these things you currently believe can only be done in the dialplan? Perhaps you have a few other misconceptions we could right for you... |
| 15:48.06 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 15:48.42 | stephendowed | @fender please what is a blf |
| 15:48.48 | [TK]D-Fender | ~blf |
| 15:48.48 | infobot | blf is probably Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
| 15:48.50 | [TK]D-Fender | ^^^ |
| 15:48.58 | stephendowed | saw that bein thrown around here |
| 15:49.15 | mjordan | [TK]D-Fender: I know you know this but... that's true if its SIP, and if the transfer is being done at the channel driver layer, i.e., the phone sends a REFER. Asterisk also does handle transfers at a higher layer using features :-) |
| 15:49.30 | [TK]D-Fender | Think of the status lights on speed dial buttons on a phone so you know if they are on a call |
| 15:50.11 | [TK]D-Fender | mjordan, Unless you're freakishly letting PSTN callers do this... then you should get a better device :) |
| 15:50.20 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 15:50.23 | stephendowed | @fender and this can also be setup to work using asterisk and "the almighty dialplan too lol" |
| 15:50.30 | [TK]D-Fender | mjordan, DTMF *-based transfers = ass :) |
| 15:50.54 | [TK]D-Fender | stephendowed, Yes, the dialplan determines what to provide BLF service for. |
| 15:51.01 | mjordan | not discussing the merits of DTMF-based transfers, just saying they can happen :-) |
| 15:51.12 | [TK]D-Fender | stephendowed, Maybe you don't want every device to be allowed to spy on the status of everyone. |
| 15:51.33 | stephendowed | @fender OIC |
| 15:51.52 | [TK]D-Fender | mjordan, Yes and like other social faux-pas' we keep them nice and buried ;) |
| 15:52.33 | [TK]D-Fender | stephendowed, OU812 ? |
| 15:53.05 | stephendowed | i mean oh i see...lol |
| 15:53.28 | [TK]D-Fender | Yes... and I returned in kind .."lol" |
| 15:54.51 | [TK]D-Fender | stephendowed, Ok, so we've covered BLF. What else? |
| 15:55.05 | stephendowed | alright whats all dese about AGI i see also in this asterisk book...not read it yet though but just gimme a heads up |
| 15:55.23 | Qwell | stephendowed: AGI = magic |
| 15:55.34 | Qwell | If Asterisk can't do it (even though it can do everything), AGI can. |
| 15:55.39 | [TK]D-Fender | stephendowed, AGI is a kind of script you call so you can call * dilaplan functionality from within a more complicated script |
| 15:55.54 | DaneoShiga | I'm having a problem with asterisk 1.8, when a call is answered by an agent on some queue, it's not calling "macro-dial" context, as happens with similar environment with asterisk 1.4... anyone could give me directions? *completely lost... |
| 15:55.59 | Qwell | I'm not really a big fan of AGI, tbh.. |
| 15:56.07 | [TK]D-Fender | stephendowed, For instance you have to do a lot of complicated DB lookups, server interaction, etc within the processing of the call and also be interactive with the user |
| 15:56.39 | [TK]D-Fender | DaneoShiga, That context isn't "special". It's existance is completely arbitrary |
| 15:57.10 | [TK]D-Fender | DaneoShiga, You would have to show us what you're actually doing along with why the dialplan is even involved in your call sitting in queue |
| 15:57.10 | Qwell | [TK]D-Fender: schroedingers context.. |
| 15:57.12 | [TK]D-Fender | ~pb |
| 15:57.12 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 15:57.16 | [TK]D-Fender | DaneoShiga, ^^^ |
| 15:57.34 | stephendowed | @fender like when u call up a bank and they can read u ur account balance? |
| 15:57.48 | Qwell | stephendowed: exactly. but, AGI isn't actually necessary for that. |
| 15:57.49 | [TK]D-Fender | Qwell, Schroedingers context? Explains ZOMBIE channels ;) |
| 15:57.55 | Qwell | [TK]D-Fender: ;D |
| 15:58.05 | [TK]D-Fender | FTMFQY! |
| 15:58.13 | [TK]D-Fender | s/Q/W/ |
| 15:58.19 | [TK]D-Fender | \o/ |
| 15:58.20 | stephendowed | so gimme a practical example u have used agi's then |
| 15:58.27 | [TK]D-Fender | stephendowed, Sure. |
| 15:58.41 | [TK]D-Fender | stephendowed, Your sample was quite common and valid |
| 15:58.51 | Qwell | stephendowed: it's a matter of preference. You can write AGIs in whatever language you like. |
| 15:58.57 | *** join/#asterisk angryuser_laptop (~angryuser@240-119-200-46.pool.ukrtel.net) |
| 15:59.14 | [TK]D-Fender | stephendowed, Or having one so an admin can call in, the scrip will check if inside IT services are active and read back states to them, etc |
| 15:59.17 | DaneoShiga | ok, gonna investigate *kind beginner with contexts and dialplans... |
| 16:00.03 | stephendowed | @fender are you serious??? so this is some kind of scripting thingy again....i hate programming hope this is fun |
| 16:00.08 | DaneoShiga | stephendowed i work with lots of agis writen in PHP, including agis for billing |
| 16:00.49 | [TK]D-Fender | stephendowed, * has some DB capabilities directly from within the dialplan. You would typically use AGI where you have comlpex tasks you can't reasonably do in the dialplan directly, or when you have just enough of the "doable but inefficient" that it's worth bundling up. |
| 16:01.13 | [TK]D-Fender | stephendowed, AGI necessisty or profitability is entirely case-based |
| 16:01.28 | [TK]D-Fender | stephendowed, It's a tool, not always the right one. |
| 16:06.50 | DaneoShiga | i think my problem hides in some freepbx stuff... |
| 16:06.58 | stephendowed | @fender nice closing line a tool not always the right one...am sure to always remember that... |
| 16:07.48 | [TK]D-Fender | DaneoShiga, .... |
| 16:07.50 | [TK]D-Fender | ~freepbx |
| 16:07.50 | 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 |
| 16:07.51 | [TK]D-Fender | ^^^^^^^^^ |
| 16:08.15 | DaneoShiga | thank you :) |
| 16:08.57 | [TK]D-Fender | stephendowed, AGI does place a certain load, so meter out its necessity or calculate if you cross a tipping point of diminishing returns. |
| 16:09.11 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 16:11.26 | *** join/#asterisk neurosys (~neurosys@c-76-108-39-221.hsd1.fl.comcast.net) |
| 16:21.38 | stephendowed | @fender aside the asterisk book what other resources can help me |
| 16:22.04 | stephendowed | i currently installed it on my vm and am playing wif it and learning new features...so how else do i go.... |
| 16:23.12 | stephendowed | and please what is 3way calling??? just saw that now |
| 16:23.32 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 16:23.59 | *** join/#asterisk Defraz (~Defraz@69.20.176.132) |
| 16:25.00 | ChannelZ | 3-way calling is when you get two lesbians on the phone with you at the same time, I think. |
| 16:25.20 | stephendowed | lmao!!!!! |
| 16:25.34 | stephendowed | so hw does that get achieved in asterisk....... |
| 16:25.40 | DaneoShiga | seems legit |
| 16:25.43 | [TK]D-Fender | stephendowed,3-way calling is also something almost always done by your phone |
| 16:26.11 | [TK]D-Fender | ChanServ, Unless you're a lesbian you're out of that deal..... |
| 16:26.35 | [TK]D-Fender | Dangit, left too soon! |
| 16:27.10 | [TK]D-Fender | stephendowed, phone supports multiple calls and bridges them for you. |
| 16:27.13 | stephendowed | so how does it work i call one of dem,hold call the other and then how do i connect the three of us...please once again am from the stone age don't crucify me |
| 16:27.42 | [TK]D-Fender | stephendowed, Most SIP phones have buttons for conferencing and 3-way, etc |
| 16:27.49 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 16:27.52 | [TK]D-Fender | stephendowed, as do other types |
| 16:28.00 | p3nguin | If someone is wanting to use his Skype on Asterisk, what options are there? In what way could he be wanting to use Skype with Asterisk? He currently has a phone number forwarding to his skype-in and he uses his skype for the phone calls. |
| 16:28.45 | ChannelZ | Well SFA is unavailable so you're stuck with Skype for SIP or whatever it's called (Skype's own product) |
| 16:29.31 | stephendowed | yeah i read they discontinued support or something like that for it.... |
| 16:29.46 | ChannelZ | http://www.skype.com/intl/en-us/business/skype-connect/ |
| 16:29.49 | stephendowed | and aside xlite what other softphones do u guys recomment.... |
| 16:30.42 | [TK]D-Fender | stephendowed, All softphone suck. Some slightly less than others |
| 16:31.02 | mentax | hi all |
| 16:31.03 | stephendowed | lol now dis is some serious hating of softphones |
| 16:31.07 | [TK]D-Fender | stephendowed, Ekiga is decent enough to start with |
| 16:31.21 | stephendowed | so which sucks less in ur opinion ekiga?? is it free? (smiles) |
| 16:31.45 | [TK]D-Fender | x-lite doesn't support transfers on the SIP side. The dialplan CAN offer them a means of doing this though. |
| 16:31.46 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 16:31.54 | [TK]D-Fender | stephendowed, X-Lite is free so that isn't the only point |
| 16:31.59 | mentax | Someone know - why when I trying to call from my sip asterisk phone it will wait around 30 sec before start call? |
| 16:32.15 | [TK]D-Fender | stephendowed, Supporting TRANSFER is kinda important. Along with other standard calling features |
| 16:32.44 | DaneoShiga | i use twinkle for testing stuff around here.. |
| 16:32.45 | [TK]D-Fender | speX-Lite is a free version from a commercial producer. Crippled accordingly to "motivate" you to buy their paid product |
| 16:35.25 | *** join/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 16:35.42 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v007-159.mobile.uci.edu) |
| 16:38.20 | p3nguin | Asterisk isn't listed as being compatible with Skype Connect. Anyone know if it really is? |
| 16:38.33 | Qwell | what is Skype Connect? |
| 16:39.30 | Qwell | it says it's just SIP |
| 16:40.20 | p3nguin | I missed that declaration. |
| 16:40.40 | Qwell | "By adding Skype Connect to your existing SIP-enabled PBX ..." |
| 16:41.28 | p3nguin | Then it is probably Skype's own SIP gateway. |
| 16:41.56 | p3nguin | But is it a tangible product, or just a service? |
| 16:42.33 | p3nguin | It looks to be a service, not a device. |
| 16:43.41 | DaneoShiga | apparently you just need to setup an sip account with skype info |
| 16:44.23 | p3nguin | I seem to rememer people here throwing around the phrases "SIP for Skype" and "Skype for SIP." I wonder if they were referring to this Skype product or something else. |
| 16:44.55 | DaneoShiga | http://www.skype.com/intl/en-us/business/skype-connect/#t_Resources |
| 16:48.40 | DaneoShiga | google says it's possible |
| 16:50.54 | p3nguin | I don't know why anyone relies on Skype for phone calls. The pricing of this service doesn't really compete with a standard ITSP's rates. |
| 16:51.04 | DaneoShiga | what doesn't work anymore is something called "skype for asterisk" |
| 16:51.14 | p3nguin | I know about SFA. |
| 16:51.26 | DaneoShiga | ah ok. |
| 16:51.29 | p3nguin | That channel has been discontinued. |
| 16:54.01 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 17:06.29 | mazpe | how can i make sure my g729 is installed properly? I see the license when i do a g729 show licenses, and my trunk to my provider is set to allow=g729 and disallow=all |
| 17:06.52 | *** join/#asterisk afink (~afink@wsip-70-184-210-123.om.om.cox.net) |
| 17:06.56 | mazpe | when i make a call sip show channels shows the call in ulaw |
| 17:10.50 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 17:13.46 | WIMPy | Maybe a dumb question, but does your provider support G.729? |
| 17:14.03 | mazpe | yes it does.. it was working on my other server. |
| 17:14.05 | *** join/#asterisk kzoo (~russellm@184.107.182.210) |
| 17:14.10 | kzoo | Does anyone here follow voiceops? If so, what's the general consensus on job postings? |
| 17:14.12 | mazpe | I even called digium to have the key release for this server |
| 17:14.25 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 17:16.09 | [TK]D-Fender | mazpe, If you restricted your peer to G.729 .... and the call is not using G.729 ... then it is not using your peer <------------- |
| 17:16.21 | [TK]D-Fender | mazpe, Now go show us configs and actual call debug |
| 17:16.27 | WIMPy | Good point |
| 17:16.51 | mazpe | ok |
| 17:16.57 | *** join/#asterisk skrusty (~ksrusty@62.252.24.138) |
| 17:18.05 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 17:19.49 | *** join/#asterisk drknus (~user@blk-222-141-162.eastlink.ca) |
| 17:21.51 | p3nguin | Maybe the allow before the disallow is causing a problem. |
| 17:22.15 | Qwell | yes, order matters |
| 17:23.20 | [TK]D-Fender | Anonymous has public struck out at a Canadian politicion tabling & supporting legislation similar to SOPA. Let the floodgates OPEN.... |
| 17:24.12 | *** join/#asterisk pdtpatr1ck (~pdtpatric@12.249.4.226) |
| 17:25.41 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:26.57 | coppice | that legislation was only started to great headlines like "Politicians wash their hands of SOPA" and "SOPA on the ropes" |
| 17:28.33 | [TK]D-Fender | Clearly |
| 17:29.19 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:30.28 | Qwell | coppice: and twitter. "What does wikipedia have against soap?!" |
| 17:30.36 | [TK]D-Fender | coppice, Lot of high positive potential volatility going on right now on the freedom front world-wide. This was one of the more visible strikes made in the western world. Middle East & Africa are getting the unfortunate ly blody side of this... |
| 17:36.24 | mazpe | [TK]D-Fender: ider support G.729? |
| 17:36.39 | mazpe | [TK]D-Fender: http://nopaste.info/a78e520199.html |
| 17:37.28 | [TK]D-Fender | mazpe, -- Executing [4042601301@default:3] Dial("SIP/208.239.76.177-00000b41", "SIP/phone_101") in new stack <- |
| 17:37.42 | [TK]D-Fender | mazpe, I do not see THIS peer which appears to be using ULAW. And I do not see the call come IN |
| 17:37.51 | [TK]D-Fender | mazpe, You aren't looking. |
| 17:37.56 | coppice | Qwell: wikipedia - the encyclopedia for the unwashed masses |
| 17:38.14 | [TK]D-Fender | mazpe, [inphonex] context=inbound |
| 17:38.21 | [TK]D-Fender | mazpe, -- Executing [4042601301@default:1] Set("SIP/208.239.76.177-00000b41", "__FROM_DID=4042601301") in new stack |
| 17:38.32 | [TK]D-Fender | mazpe, does that look like "inbound"to you? |
| 17:38.41 | mazpe | i see |
| 17:38.57 | [TK]D-Fender | mazpe, Now go actually loko at the call |
| 17:39.00 | [TK]D-Fender | look* |
| 17:39.10 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-bgrqjbxvxmhixeqb) |
| 17:40.39 | *** join/#asterisk bindi (~bindi@f430.ip15.netikka.fi) |
| 17:41.09 | bindi | hey, i've got asterisk 1.6.2.9-2+squeeze4, how can i send and receive SMS? |
| 17:41.28 | bindi | just says "warning, receiveed message to .. by .., dropped it" |
| 17:42.48 | [TK]D-Fender | bindi, * is not an SMS platform |
| 17:43.07 | bindi | right well shouldn't it be possible anyway? |
| 17:43.19 | bindi | calls and sms would be super-awesome, can't be calling all the time |
| 17:43.59 | [TK]D-Fender | bindi, A lot of things would be "awesome". This one is not happening. |
| 17:44.44 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:45.16 | bindi | how come? |
| 17:46.10 | bindi | do you know any software that lets me do that then? |
| 17:46.25 | [TK]D-Fender | bindi, Same reason bikes don't fly: Doesn't have wings |
| 17:46.45 | [TK]D-Fender | bindi, Why isn't * also a word processor? because no-one added that functionality. |
| 17:47.16 | p3nguin | I'd think app_sms would be able to do something. |
| 17:47.21 | [TK]D-Fender | bindi, Doesn't matter if the concept is related. It simply isn't supported yet. |
| 17:47.34 | [TK]D-Fender | p3nguin, Send-only an E1... |
| 17:47.44 | [TK]D-Fender | And limited usability. |
| 17:48.27 | p3nguin | That's a pretty crappy limitation. People outside of the UK and Italy like to use SMS, too. |
| 17:48.59 | mazpe | [TK]D-Fender: interesting i just noticed the following as well [Feb 23 12:48:22] WARNING[7151]: translate.c:808 __ast_register_translator: Invalid translator path: (starting codec is not valid) |
| 17:49.30 | p3nguin | How about showing the configuration like you were asked? |
| 17:49.32 | [TK]D-Fender | p3nguin, * also doesn't talk SCCP as a phone for server>server SCCp in Cisco environments. * doesn't transcode or mix video for conferencing, and lots of other stuff as well. |
| 17:49.47 | [TK]D-Fender | mazpe, I'm not see what I requested... |
| 17:51.22 | mazpe | yup... brb |
| 17:59.34 | *** join/#asterisk qakhan (~qakhan@180.178.151.59) |
| 17:59.39 | qakhan | hi all |
| 18:00.35 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:00.35 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:01.17 | qakhan | i have * in my local network. i setup port forwarding on router to * for register exts from internet. exts get register but i cannot hear caller voice |
| 18:01.31 | [TK]D-Fender | ~sipnat |
| 18:01.31 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 18:01.36 | qakhan | is there any thing which i missed |
| 18:01.38 | [TK]D-Fender | qakhan, ^^^^^^^ |
| 18:01.48 | [TK]D-Fender | qakhan, Read the guide. |
| 18:02.06 | [TK]D-Fender | qakhan, And in 1.6+ "canreinvite" has been replaced by "directmedia". |
| 18:02.18 | p3nguin | externip replaced with externaddr |
| 18:02.22 | qakhan | i m using 1.4.38 |
| 18:02.48 | p3nguin | Then those terms are correct. |
| 18:02.53 | [TK]D-Fender | qakhan, then the guide applies to you as-written |
| 18:07.27 | adeel|work | anyone happen to have a tool to convert SIP log messages to a pcap file or better analysis tool? |
| 18:08.07 | [TK]D-Fender | adeel|work, mat tcpdump |
| 18:08.11 | [TK]D-Fender | adeel|work, man tcpdump |
| 18:09.02 | adeel|work | [TK]D-Fender, tcpdump can take just the application level SIP messages and convert it? interesting... |
| 18:12.08 | p3nguin | Maybe wireshark would be easier. |
| 18:12.45 | adeel|work | i can't seem to import the text file into wireshark |
| 18:13.34 | *** join/#asterisk erth64net (~tocici@pdxvmh15.tocici.com) |
| 18:14.09 | p3nguin | What file are you talking about? |
| 18:14.45 | Kobaz | okay so |
| 18:15.10 | Kobaz | anyone ever have a problem with MixMonitor when the legs of the call are overlapping each other |
| 18:15.32 | Kobaz | the audio in the recording does not match the actual audio that was sent/received |
| 18:18.23 | Kobaz | here's an example |
| 18:18.30 | Kobaz | http://www.kobaz.net/misc/1329956043.4232070.WAV |
| 18:18.38 | Kobaz | the recording has people talking over each other |
| 18:18.47 | Kobaz | but that didn't actually happen on the real phone call |
| 18:19.31 | WIMPy | Does anyone have some experience with Asterisks H.323 implementation? How good / stable / complete is it? |
| 18:20.04 | pabelanger | WIMPy: chan_h323 isn't maintained. |
| 18:20.10 | pabelanger | chan_ooh323 has somebody |
| 18:21.23 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 18:23.56 | Kobaz | the weird thing is that it's not every single call |
| 18:26.59 | Dovid | i got cepstral for asterisk and get the errorARNING[11041]: loader.c:387 load_dynamic_module: Error loading module 'res_cepstral.so': libswift.so.5: cannot open shared object file: No such file or directory |
| 18:27.04 | Dovid | anyone know what i am misisng? |
| 18:27.24 | *** join/#asterisk erth64net (~tocici@pdxvmh14.tocici.com) |
| 18:27.34 | kzoo | Dovid, Make sure /opt/swift/lib is in your /etc/ld.so.conf, then run /sbin/ldconfig |
| 18:30.34 | Dovid | thanks |
| 18:31.58 | *** join/#asterisk hfb (~hfb@pool-98-112-209-179.lsanca.dsl-w.verizon.net) |
| 18:33.21 | *** join/#asterisk _pepo_ (c837e07d@gateway/web/freenode/ip.200.55.224.125) |
| 18:33.24 | _pepo_ | hi friends |
| 18:33.41 | Dovid | whats the diff between say text and say speach? |
| 18:36.42 | _pepo_ | I have two asterisk interconected... when I call from one extension in first box to another extension in second box and the second is in fail, What message should I get in response: 5XX or 4XX? |
| 18:36.57 | *** part/#asterisk davidduffy (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 18:37.29 | Kobaz | depends how it failed |
| 18:37.44 | Kobaz | the message you get back will be based on what went wrong |
| 18:39.29 | _pepo_ | What if the line is shorted? |
| 18:40.29 | WIMPy | ][ |
| 18:40.31 | WIMPy | ] |
| 18:40.43 | WIMPy | Oops. Sorry. |
| 18:40.52 | _pepo_ | I means: if receive message 5XX (502) I can suposse that the server is wrong but is just one extension... |
| 18:44.30 | *** join/#asterisk davidduffy (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
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| 19:21.19 | butchevans2001 | I have installed chan_sccp_b on asterisk 1.8.9.2. I am trying to get the sccpdevice and lines via mysql. When asterisk starts, mysql is no seeing a mysql query happening and consequently, no devices are loaded. Is there an easy way to test (from asterisk) why this is happening? |
| 19:23.49 | Qwell | Did they ever actually add realtime support to it? |
| 19:24.09 | butchevans2001 | documentation indicates that it is there |
| 19:24.30 | butchevans2001 | behavior makes me think maybe it isn't |
| 19:27.35 | [TK]D-Fender | butchevans2001, Have you considered showing backup? |
| 19:31.20 | p3nguin | What's the easiest way to check and monitor bandwidth usage of some phone calls? |
| 19:31.31 | butchevans2001 | showing backup? |
| 19:34.04 | Qwell | p3nguin: I like mrtg |
| 19:34.29 | p3nguin | You monitor the router with snmp? |
| 19:35.38 | p3nguin | I'll see if I can make that work in my situation. I was kind of hoping for some type of real-time graph or chart software that can watch an interface on the asterisk system and say what it is using at any given moment. |
| 19:37.19 | p3nguin | Oh, I just had a primitive idea: I can use ifconfig to check the bytes in/out, wait a certain amount of time, check it again, and do the math. |
| 19:39.29 | butchevans2001 | p3nguin: iptraf should give you what you want |
| 19:39.41 | butchevans2001 | if you want realtime view of traffic |
| 19:53.38 | *** join/#asterisk dxd828 (~dxd828@88-104-64-87.dynamic.dsl.as9105.com) |
| 19:53.51 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 19:57.50 | *** part/#asterisk dxd828 (~dxd828@88-104-64-87.dynamic.dsl.as9105.com) |
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| 20:15.03 | butchevans2001 | [TK]D-Fender: I don't understand what you mean by "Have you considered showing backup?" |
| 20:16.17 | [TK]D-Fender | butchevans2001, You haven't shown us any of your configs, CLI output from load attempts, DB lookups to validate that the credentials are right, etc. So far all we have is an Asterisk version number and "it doesn't work". |
| 20:18.18 | *** join/#asterisk TysonBrooks (me@c-24-2-91-162.hsd1.ut.comcast.net) |
| 20:19.01 | butchevans2001 | ahh. I'm more looking for documentation/assistance toward how I can see what asterisk is actually doing. What I DO know is that asterisk is NOT talking to mysql to get the information. I'm not asking for a "tell me what to configure.." as much as "how can I troubleshoot". I am wanting to learn something from this.. |
| 20:19.57 | butchevans2001 | I will post the relevant configs in a bit...just have to clean them up for passwords |
| 20:20.53 | [TK]D-Fender | butchevans2001, troubleshooting : look at all configs for errors. make sure system is even looking at those configs. Make sure that what you're asking them to do is matching up with external resources |
| 20:21.13 | butchevans2001 | also, if it matters, I have validated the user/pass with mysql and can connect and view the tables via mysql cli with the user/pass |
| 20:21.28 | TysonBrooks | What is the recommended flavor of linux to run asterisk on? |
| 20:21.46 | Qwell | TysonBrooks: strawberry |
| 20:21.48 | butchevans2001 | Is there an asterisk command that would tell me if asterisk is using the files? |
| 20:22.00 | butchevans2001 | err...looking at the configs |
| 20:22.20 | Qwell | TysonBrooks: Most of the Digium developers use Debian or some variant. It really doesn't matter though. Use what you know. |
| 20:22.26 | TysonBrooks | I'm interested in building an asterisk box and using it for google voice for my office. |
| 20:22.48 | butchevans2001 | I know for sure that asterisk is not sending any queries to mysql asking for the sccplines/sccpdevices tables |
| 20:26.36 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 20:30.41 | autofsckk | hello, im testing with a vps, i connect it with my service provider, i can make calls, but im not receiving them, i have the same config that i was using with my laptop and it was working great, where should i start to see where the problem is? |
| 20:31.03 | Qwell | are you registering? |
| 20:31.07 | Qwell | firewall open? |
| 20:32.12 | [TK]D-Fender | See any attempts coming in? |
| 20:32.20 | [TK]D-Fender | Who's on first? |
| 20:32.24 | [TK]D-Fender | Who shot J.R.? |
| 20:32.38 | [TK]D-Fender | What is the average airspeed velocity of an unladen swallow? |
| 20:36.00 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 20:38.01 | butchevans2001 | [TK]D-Fender: swallow velocity is here: http://style.org/unladenswallow/ |
| 20:46.20 | Kobaz | 42 |
| 20:48.10 | autofsckk | so sorry i needed to move, im back here |
| 20:48.30 | autofsckk | [TK]D-Fender: no attempts coming |
| 20:48.56 | autofsckk | Qwell: yes im registered, i opened udp 5060 and 10000-3000 |
| 20:50.18 | [TK]D-Fender | autofsckk, pastebin a registration attempt |
| 20:53.08 | autofsckk | [TK]D-Fender: sorry for my ignorance in advance, but how can i make a registration attempt? i see with sip show registry that im already registered to my providers server |
| 20:53.21 | [TK]D-Fender | sip reload |
| 20:53.42 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 20:53.55 | autofsckk | ok |
| 20:54.44 | autofsckk | and where do i take the paste from? i got just 3 lines from the cli, let me paste them, give me a sec |
| 20:55.37 | *** join/#asterisk dxd828 (~dxd828@88-104-64-87.dynamic.dsl.as9105.com) |
| 20:55.53 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:56.35 | *** part/#asterisk dxd828 (~dxd828@88-104-64-87.dynamic.dsl.as9105.com) |
| 20:57.56 | *** join/#asterisk darkskiez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 20:58.11 | autofsckk | http://paste2.org/p/1914015 sorry pastebin.com isnt working |
| 20:58.17 | autofsckk | you mean that paste [TK]D-Fender ? |
| 20:59.01 | [TK]D-Fender | auoShows you clearly haven't enabled SIP DEBUG before doing this |
| 20:59.11 | [TK]D-Fender | autofsckk, Shows you clearly haven't enabled SIP DEBUG before doing this |
| 20:59.43 | [TK]D-Fender | autofsckk, please do so and repeat this process |
| 21:07.30 | *** join/#asterisk twodogs (~twodogs@hendra.biohazard.seattle.wa.us) |
| 21:13.50 | *** join/#asterisk darkskiez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 21:16.51 | p3nguin | Why does a call using ulaw/64k use 90 kbit/s per call leg? |
| 21:17.03 | p3nguin | Is there just that much overhead in the packets? |
| 21:17.19 | Qwell | p3nguin: 80, and yes |
| 21:17.36 | p3nguin | The other 10 must be for session control. |
| 21:17.45 | WIMPy | Depending on packetization. |
| 21:17.46 | Qwell | shouldn't be |
| 21:17.58 | WIMPy | And RTCP |
| 21:22.37 | *** join/#asterisk _jdccdevel (~chatzilla@69.196.87.218) |
| 21:29.36 | *** join/#asterisk fornax (~fornax@85.183.53.64) |
| 21:30.09 | *** join/#asterisk citywok (~kvirc@67-134-194-33.dia.static.qwest.net) |
| 21:42.10 | *** join/#asterisk fornax (~fornax@85.183.53.64) |
| 21:42.14 | fornax | is someone here that could help me with an asterisk installation that produces tons of segfaults when I use the asterisk caldav support? |
| 21:42.37 | Qwell | What version of Asterisk? |
| 21:43.15 | fornax | I have the issue from 1.6 - 10.1 currently running 10.1 on gentoo |
| 21:51.20 | *** join/#asterisk fornax (~fornax@85.183.53.64) |
| 21:51.25 | fornax | re, my system crashed, i always have to remove the coredumps because my hd is full within minutes |
| 21:51.39 | fornax | did you write something, sorry |
| 21:52.13 | Qwell | ~backtrace |
| 21:52.13 | infobot | backtrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt). See also https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 21:52.42 | *** join/#asterisk jaybee_ (~quassel@114.31.212.161) |
| 21:53.06 | fornax | i already opened this bug and added backtraces https://issues.asterisk.org/jira/browse/ASTERISK-19346 |
| 21:53.51 | Qwell | Those aren't backtraces... |
| 21:53.53 | fornax | but i think the backtraces are not very useful because information is lacking. I recompiled my whole gentoo with all possible debug flags but I cannot get better backtraces then my submitd ones |
| 21:53.54 | Qwell | You missed a few steps. |
| 21:54.19 | jaybee_ | Hi all. Is there any way, in asterisk [dialplan], to check if extension ${SOMEEXT} is logged into queue ${SOMEQUEUE}? |
| 21:55.48 | fornax | Qwell: So gee me a hint what is missing. I did exactly all steps from the manual |
| 21:56.06 | Qwell | ...the backtrace part |
| 21:57.09 | fornax | Qwell: I recompiled asterisk with better backtraces and dont_optimize and then used all steps from the page that you sent me |
| 21:58.27 | fornax | i would really like to add more information to the thread and invested hours but I really don't know what is still missing |
| 21:59.18 | Qwell | fornax: all you did was load gdb, then quit |
| 21:59.44 | fornax | i did this to generate a backtrace file: gdb -se "asterisk" -ex "bt full" -ex "thread apply all bt" --batch -c core-20120221-145828 > /tmp/backtrace.txt |
| 21:59.57 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 22:00.20 | fornax | and then I did: gdb -se "asterisk" -c core-20120221-145828 | tee /tmp/backtrace.txt |
| 22:01.25 | fornax | and there is no more information then the posted backtrace.txt files. I can do this for all backtraces that were produced by ast |
| 22:02.57 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:03.08 | Qwell | and the second one overwrote the file that you created with the first. |
| 22:03.51 | fornax | but this is exactly what is written on the webpage |
| 22:04.49 | fornax | but you are right, the backtrace before is far better |
| 22:05.48 | *** join/#asterisk jpsharp (jsharp@ohno.mrbill.net) |
| 22:06.13 | fornax | i will upload the one before i applied the first command. I think the webpage is misleading |
| 22:09.58 | jpsharp | I'm using realtime sippeers, most of my peers work right, but there's a few that don't show up via "sip show peers", but do show up with "Realtime load sippeers". |
| 22:10.18 | Qwell | jpsharp: sip show peers doesn't get everything from the database - it only gets what it knows about |
| 22:10.22 | fornax | Qwell: I uploaded a correct backtrace. |
| 22:10.40 | Qwell | fornax: much better |
| 22:11.30 | fornax | Qwell: Sorry, that I was too silly and spammed the issue with crap |
| 22:11.36 | jpsharp | Qwell: All my other realtime-loaded peers show up via sip show peers. What could be different about these? |
| 22:11.52 | Qwell | jpsharp: They have been accessed in some way already. |
| 22:12.23 | jpsharp | They won't show up until something tries to register to them? |
| 22:12.32 | Qwell | or use them in some way |
| 22:12.50 | Qwell | sip show peer <somepeerthatisntloadedyet> |
| 22:13.12 | jpsharp | The dialer script tries to place a call to that peer and Asterisk ays it knows nothing about the peer. |
| 22:13.36 | jpsharp | Lemme register something against it & see if that does it. |
| 22:13.46 | Qwell | If it isn't registered, of course it doesn't know how to contact it. |
| 22:14.01 | jpsharp | It just says "Not found" |
| 22:14.18 | jpsharp | as though it never existed in sip.conf |
| 22:16.26 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-xabrwmvgvalbpaji) |
| 22:16.26 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 22:16.49 | jpsharp | I'm troubleshooting a DID. I need to see if it is a problem with the * config or PEBKAC. |
| 22:19.17 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:20.35 | jpsharp | I can see it via show peers once I try to register against it. |
| 22:20.52 | jpsharp | So the end user isn't registering right, ergo PEBKAC. |
| 22:24.08 | fornax | Qwell: Can you see something when you read the backtrace? |
| 22:24.24 | Qwell | fornax: I'm not looking at that issue. I was just confirming that your backtrace is now correct. |
| 22:25.19 | fornax | Qwell: okay, thank you, then I have to wait until the maintainer will have a look. Who will update the status flag of the issue? |
| 22:30.47 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 22:41.06 | autofsckk | finally fixed the no getting calls :D |
| 22:41.09 | autofsckk | damned hehehe |
| 22:42.02 | autofsckk | in the sip.conf i used the username=whatever it was different from the head part of the trunk info |
| 23:05.20 | *** join/#asterisk blitzrage (~androirc@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:05.21 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 23:26.03 | *** join/#asterisk scgm11 (~Sebastian@r186-52-27-157.dialup.adsl.anteldata.net.uy) |
| 23:27.45 | scgm11 | one question, is asterisk 10 production ready? I´m planning to move from 1.6.2.X to another version, I´m evaluating 1.8 branch and 10 but LTS seems a big deal, with the small support windows for asterisk 10 do you think you are having many people testing it?? |
| 23:29.10 | TysonBrooks | Whats the difference between asterisk and askterisk@home? |
| 23:31.45 | scgm11 | there is now more asterisk home, the name is trixbox, basically is a distribution that has his own GUI |
| 23:33.05 | TysonBrooks | I guess what I'm looking at is the AsteriskNOW and Asterisk, whats the differences? |
| 23:34.25 | scgm11 | again is a distribution, a pbx with a gui, the main difference here is that asterisk now if a digium product |
| 23:35.09 | TysonBrooks | Which would be the way to go if I want to just run a VOIP solution for Google Voice? |
| 23:37.07 | [TK]D-Fender | same thing |
| 23:37.21 | [TK]D-Fender | TysonBrooks: AsteriskNOW is a complete distro that INCLUDES Asterisk |
| 23:37.42 | [TK]D-Fender | TysonBrooks: Your question is akin to "what is better, a car or a wheel?" |
| 23:39.26 | TysonBrooks | Sorry, okay let me go this route... Last night I was watching the webinar thats listed on the website, and in there it was saying something about Google Voice. Am I able to setup Asterisk to use Google Voice's service as my telephone number? |
| 23:39.53 | [TK]D-Fender | yes |
| 23:40.04 | [TK]D-Fender | however it is a poor business decision. |
| 23:40.12 | TysonBrooks | how so? |
| 23:40.20 | [TK]D-Fender | Their terms of service could change at any time and you'll find yourself stranded |
| 23:40.33 | TysonBrooks | I'm just doing this in a sandbox vm right now anyway. |
| 23:40.46 | [TK]D-Fender | And there are occasionally breakages in thier protocols and * has had issues |
| 23:41.33 | *** join/#asterisk tmrhmdv (457279e9@gateway/web/freenode/ip.69.114.121.233) |
| 23:41.48 | TysonBrooks | I was just looking for a solution that was next to nothing right now. My current phone system consits of gvoice, skype and my cell phone. |
| 23:42.06 | TysonBrooks | costing roughly $30 every 3 months with out my cell phone in the mix |
| 23:42.30 | tmrhmdv | Hello everyone! Do I have to pay for using the G729 codec as someone buying traffic from a SIP wholesaler? When and why should people pay for G729? |
| 23:42.50 | [TK]D-Fender | tmrhmdv: You need to the moment you have to transcode anything |
| 23:43.58 | scgm11 | you can use g729 passthrough though, if both endpoint has g729 you need no license |
| 23:46.03 | tmrhmdv | If I am not doing any trans-coding, I don't have to pay anything? |
| 23:46.33 | tmrhmdv | scgm11: Does * support G729? I heard that I have to buy lic. for each channel? |
| 23:47.14 | [TK]D-Fender | Correct |
| 23:47.41 | [TK]D-Fender | You need a license for every channel that will be simultaneously transcoding |
| 23:48.06 | tmrhmdv | Aah, it's clear now. Thank you very much! |
| 23:50.10 | TysonBrooks | You need more space on the following file systems: 286 M on / ---- is this disk space or what? |
| 23:52.58 | [TK]D-Fender | yes |
| 23:53.04 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 23:53.21 | TysonBrooks | Whats the recommended hard drive space/ |
| 23:54.18 | [TK]D-Fender | Enough so it doesn't complain and you don't run out |
| 23:54.39 | TysonBrooks | I had 2.5Gig allocated for it. |
| 23:55.56 | [TK]D-Fender | well you are getting very clear warnings. I'd advise doing something about those.... |
| 23:56.22 | *** join/#asterisk Defraz (~Defraz@69.20.176.132) |
| 23:59.41 | ketas | does grandstream gxp2000 suck? |