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| 00:17.42 | Sprocks | how do I get voicemail to use callerid name as the mailbox? |
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| 00:24.35 | [TK]D-Fender | Sprocks: pass it in your app call. |
| 00:24.48 | Sprocks | someone suggested VoiceMailMain(${CALLERID(name)}@VoiceMail) but that doesn't work for me |
| 00:25.28 | [TK]D-Fender | And is that the VM context you use? |
| 00:25.47 | [TK]D-Fender | Also.... the NAME is generally TEXT |
| 00:25.54 | [TK]D-Fender | VM box are supposed to be NUMERIC |
| 00:26.37 | Sprocks | in my voicemail I have name => pass,name,email |
| 00:27.06 | [TK]D-Fender | VM is not supposed to support "named" boxes, only numeric |
| 00:28.56 | Sprocks | ok so I should use the extension used to dial the user as their mailbox id? |
| 00:29.43 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 00:30.17 | [TK]D-Fender | Sprocks: Repeat after me : Box NUMBER ....... |
| 00:40.19 | WIMPy | Mailbox names don't seem to be restricted to be numberic. |
| 00:40.38 | WIMPy | But they are (or were) resticted in length. |
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| 03:23.33 | ChannelZ | <off topic> iTunes is a pile of shit. </off topic> |
| 03:24.51 | Maliuta | ChannelZ: <off topic> no shite sherlock </off topic> |
| 03:52.09 | ChannelZ | Just in case someone didn't know. |
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| 04:08.24 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 04:26.25 | youjelly | >apple |
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| 06:37.03 | gavimobile | member=SIP/1234567@my-trunk calls my cellphone number from a queue. It comes in as a blocked number though, how can I apply Set(CALLERID(num)=${CALLERIDVAR} |
| 06:37.40 | gavimobile | any help would be appreciated |
| 06:42.14 | *** join/#asterisk jsjc (~Adium@213.37.155.15) |
| 06:47.11 | kaldemar | what is CALLERIDVAR? |
| 06:49.20 | kaldemar | and what calls what really? members don't call anything, they are members of a queue that get called when a queue gets a caller. |
| 06:50.32 | kaldemar | is it your cellphone that makes the call, it ends up in a queue that has the mentioned number, but the member can't see a caller id? |
| 06:51.11 | kaldemar | s/mentioned number/mentioned member/ |
| 06:52.14 | gavimobile | kaldemar: it's a variable I created which contains my caller id |
| 06:53.08 | gavimobile | kaldemar: when I add that line, member=SIP/1234567@my-trunk in queues.conf I get a phone call from a private number when someoen calls the queue |
| 06:53.33 | gavimobile | call it whatever you want in termonology, but how can I acomplish what im trying to do |
| 06:53.39 | gavimobile | which is call my cellphone with a specific number |
| 06:54.19 | kaldemar | i'm trying to figure out what happens, not bully you about terminology. |
| 06:54.32 | gavimobile | kaldemar: haha! |
| 06:54.41 | gavimobile | kaldemar: I thought you possibly didn't understand me |
| 06:54.44 | kaldemar | how does the call come in to your asterisk box? |
| 06:54.57 | gavimobile | via sip? |
| 06:55.01 | gavimobile | is that what your asking |
| 06:55.09 | kaldemar | partly. |
| 06:55.26 | gavimobile | well than I didn't understand the question |
| 06:55.54 | kaldemar | you probably shouldn't set the caller is to a constant value on all incoming calls. |
| 06:56.09 | kaldemar | is the call comin from a provider? |
| 06:56.45 | kaldemar | or do you have a sip client of some sort in your cell phone? |
| 06:57.27 | gavimobile | kaldemar: I do have a sip client but I want the call to come via gsm |
| 06:57.43 | gavimobile | in english terms I believe we call it followme |
| 06:58.31 | gavimobile | not sure what you meaan by coming from a provider. the call is coming from someone who is calling my main phone number which than is added to the queue |
| 06:59.09 | gavimobile | I think one way of doing this is with a virtual extention which calls myt cellphone |
| 06:59.27 | gavimobile | just not sure how to do it |
| 07:02.11 | kaldemar | provider as in an ITSP that gives you a number in PSTN and forwards calls to that number to you via SIP. |
| 07:02.45 | kaldemar | so how does the call come in now? |
| 07:03.28 | gavimobile | no I don't think I want that.. I simply want that If I leave the office and someone calls than the call is still routed to my office, just the office forwards the call from me. I don't want the itsp to forward me the call |
| 07:03.51 | gavimobile | the itsp forwards the call to my pbx |
| 07:04.50 | kaldemar | and is there no caller id anywhere in the SIP INVITE? |
| 07:05.18 | gavimobile | kaldemar: I don't need the caller id of the caller |
| 07:05.38 | gavimobile | I want the caller id of my office so I know my office is forwarding a call to me |
| 07:06.04 | gavimobile | not sure what you ment |
| 07:06.29 | gavimobile | I know for a fact I can control the caller id of my channels from my did provider |
| 07:06.36 | gavimobile | I can "spoof" shall we call it |
| 07:06.50 | kaldemar | set it in a place of your choice in your dialplan then. |
| 07:16.10 | gavimobile | kaldemar: I tried but it didn't work |
| 07:16.15 | gavimobile | could you possibly show me an example |
| 07:16.17 | gavimobile | ? |
| 07:16.59 | gavimobile | say if I call user 0000FFFF0005 which we will say exten => 105,1, how can I have this transfer the call to my cell |
| 07:43.00 | *** join/#asterisk mathi (~Matthew@109.133.242.136) |
| 07:43.02 | mathi | hi |
| 07:44.15 | mathi | How can I get the number of the user and tell that number to him. The calls are coming from PSTN |
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| 07:50.43 | youjelly | calllerid(num) |
| 07:53.17 | mathi | youjelly, thanks, how would you tell it back to the user, one by one? I would need to split the numbers and loop with playback each one of them |
| 07:53.37 | mathi | but is there anyway to split numbers? |
| 07:54.05 | youjelly | you could use some tts like |
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| 07:54.49 | mathi | tts? |
| 07:55.01 | youjelly | or rather http://www.voip-info.org/wiki/view/Asterisk+cmd+SayDigits |
| 07:55.22 | mathi | I'll give that a try |
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| 08:23.20 | kaldemar | gavimobile: exten => 105,1,Set(CALLERID(num)=${CALLERIDVAR}) same => n,Dial(SIP/0000FFFF0005) |
| 08:23.48 | kaldemar | gavimobile: if it goes through a queue, you'd have Queue(...) instead of Dial(SIP/...) |
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| 09:07.11 | mathi | I can't make curl to work: |
| 09:07.17 | mathi | same => n,Set(myUrl=http://www.example.be/ivr/app/get) |
| 09:07.27 | mathi | same => n,Set(myCurlData=${CURL(${myUrl})}) |
| 09:07.32 | mathi | same => n,Verbose(1, CURL data: ${myCurlData}) |
| 09:07.59 | mathi | it returns empty string for me, but there is a string returned by my site (replace my site name by example) |
| 09:08.09 | mathi | *replaced |
| 09:09.08 | mathi | (curl is installed and I can see in the logs that libcurl is usable) |
| 09:09.52 | Sprocks | how do I solve "pbx.c:3810 ast_func_read: Function selected not registered" |
| 09:13.48 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 09:15.16 | *** join/#asterisk danfromuk (~IceChat77@2.27.38.64) |
| 09:15.47 | danfromuk | Has anyone integrated Google Analytics with Asterisk before? |
| 09:22.56 | ChannelZ | mathi: double check your URL that you don't have a typeo and that it really is returning something if you wget it or hit it in a browser |
| 09:25.17 | ChannelZ | Sprocks: I'm guessing because you are a calling a function that doesn't existt |
| 09:26.00 | mathi | ChannelZ, I am 100% sure, I copy paste the URL from the code put it in browser, and I get a result. also I installed libcurl and ./configure tells it can use it. I also tried to make/make install again. I have no idea now |
| 09:26.24 | Sprocks | ChannelZ I'm trying to set up a menu |
| 09:28.27 | ChannelZ | Sprocks: ok that tells me nothing |
| 09:29.19 | ChannelZ | mathi: do you have access to the webserver logs to see if Asterisk is even making the request? It works for me. |
| 09:29.36 | Sprocks | ChannelZ, call in get a menu telling you what number to push for people, push number, dial person |
| 09:30.07 | Sprocks | i found a tutorial online that is for 1.6 and didn't seem to work |
| 09:30.58 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 09:31.23 | ChannelZ | well without seeing some dialplan or console output I don't know what to tell you |
| 09:32.11 | mathi | ChannelZ, I don't have web server logs :( for example http://icanhazip.com doesn't work either, so it's not a problem related to the URL |
| 09:33.54 | ChannelZ | mathi: try http://burner.com/fart |
| 09:34.55 | Sprocks | ChannelZ http://pastebin.com/sNjWZz08 |
| 09:35.41 | mathi | ChannelZ, I just tried, it didn't work. Is it your server ? |
| 09:36.07 | mathi | but I accessed it once with a chrome browser (in case it would be your server) |
| 09:36.32 | ChannelZ | Yes, and I didn't see a request. |
| 09:36.54 | mathi | you saw my chrome request? |
| 09:36.56 | ChannelZ | I mean I saw your browser one from Chrome |
| 09:36.59 | mathi | ok |
| 09:37.01 | ChannelZ | but nothing prior to that |
| 09:37.10 | mathi | strange |
| 09:37.17 | mathi | what shall I do ? |
| 09:37.42 | ChannelZ | can you 'wget' from that same machine? Is it blocked by iptables, or something else going on... I'm not sure |
| 09:39.12 | ChannelZ | Sprocks: the syntax on your GotoIf is messed up. |
| 09:39.37 | mathi | ChannelZ, http://pastebin.com/YLQNugUZ |
| 09:40.01 | ChannelZ | GoToIf($[${selected}=1]?dialone) |
| 09:40.37 | mathi | ChannelZ, there is a 200 OK response, so you must have seen it ? |
| 09:40.41 | ChannelZ | mathi: well you have no write access to wherever you tried from but other than that I saw the request |
| 09:41.18 | mathi | ChannelZ, so it's an issue with my CURL function in asterisk ... |
| 09:41.22 | mathi | it's not working :( |
| 09:44.27 | mathi | ChannelZ, i will try to type curl from the console, not from asterisk |
| 09:45.11 | mathi | ChannelZ, yep from the console i get the string |
| 09:45.22 | mathi | so curl works from command, but not from asterisk |
| 09:45.29 | mathi | *console |
| 09:46.32 | Sprocks | thanks ChannelZ |
| 09:47.42 | ChannelZ | sure.. btw the easier way to do that is to just make an extension called 1 or whatever. |
| 09:49.22 | Sprocks | i was thinking of having it ring multiple people thats why I chose this way, plus it was the tutorial I found |
| 09:50.40 | Sprocks | also I was gonna have a Record(/tmp/newrec:gsm) extension but I can't find the recordings after it claims to have made them |
| 09:50.55 | ChannelZ | Well unless you're reading the digit and then needing to do a bunch of other processing based on what they hit, there's no real reason to use Read() and then build a bunch of crazy GotoIfs |
| 09:51.48 | ChannelZ | But either way |
| 09:52.59 | ChannelZ | And unless that was just a typeo, it'd be newrec.gsm not newrec:gsm |
| 09:53.08 | mathi | ChannelZ, do you have any idea how I could fix the issue? |
| 09:53.11 | ChannelZ | Plus whatever user your Asterisk runs as would obviously need write access to /tmp |
| 09:54.29 | ChannelZ | mathi: no, not really sure what could be happening off the top of my head. I assume you have libcurl and not just the libcurl-dev package |
| 09:55.20 | mathi | ChannelZ, i think i installed libcurl-dev and curl |
| 09:56.00 | ChannelZ | libcurl-dev should have also gotten libcurl (or libcurl3 or somesuch) as a dependency |
| 09:56.57 | ChannelZ | Unless you're getting an error in the console of some sort, I'm not sure what is going on. |
| 09:57.31 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 09:58.06 | mathi | ChannelZ, is there anyone I can contact that could help me? |
| 09:58.41 | mathi | maybe I should file a bug |
| 09:58.48 | ChannelZ | pastebin your console output with verbose on |
| 09:58.52 | ChannelZ | And what version of * are you using |
| 10:00.33 | mathi | ChannelZ, give me a sec:) |
| 10:06.35 | *** join/#asterisk Kraln (~kraln@69.169.90.239) |
| 10:09.13 | mathi | ChannelZ, asterisk-1.8.7.1 http://pastebin.com/raw.php?i=cCZr6Wbv |
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| 10:11.13 | ChannelZ | hmm |
| 10:11.31 | ChannelZ | and post your exact dialplan for that |
| 10:13.02 | mathi | ChannelZ, http://pastebin.com/raw.php?i=8WTtrfUJ |
| 10:13.32 | ChannelZ | so much for 'exact' but OK |
| 10:14.42 | mathi | the two main instructions are: |
| 10:14.42 | mathi | same => n,Set(myCurlData=${CURL(${myUrl})}) |
| 10:14.48 | mathi | same => n,Verbose(1, CURL data: ${myCurlData}) |
| 10:14.53 | ChannelZ | Not sure what to say, it works here. |
| 10:15.14 | mathi | what version of asterisk do you have ? |
| 10:15.23 | ChannelZ | 10.7.0 |
| 10:15.40 | mathi | such a different version ... |
| 10:15.48 | ChannelZ | you never told me what you were on |
| 10:15.54 | mathi | ChannelZ, asterisk-1.8.7.1 |
| 10:15.58 | mathi | I said before |
| 10:16.04 | mathi | I'm on ubuntu |
| 10:16.28 | ChannelZ | oh yeah this is a package or did you rebuild from source? |
| 10:16.56 | mathi | ChannelZ, to install asterisk I did exactly what the docs suggested |
| 10:17.11 | mathi | with make, make install, etcetera |
| 10:17.43 | mathi | maybe I should like to install another version, but why is your numbering so different than mine ? |
| 10:18.22 | ChannelZ | 1.8.7 is a little outdated but I don't know specifically about a bug |
| 10:18.24 | *** join/#asterisk Kraln (~kraln@69.169.90.239) |
| 10:18.38 | ChannelZ | I'm running Asterisk 10. It's what came after 1.8 |
| 10:18.46 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 10:22.14 | ChannelZ | unfortunately I don't have any machines running 1.8 to test and see if it's broken for some reason |
| 10:22.54 | mathi | ChannelZ, I will install another version, i need some time now though :P |
| 10:23.03 | ChannelZ | Did you put in libcurl3 or 4? |
| 10:23.42 | mathi | 4 as suggested in the docs |
| 10:26.46 | mathi | ChannelZ, instead of 1.8 in all these instructions: cd ~/src/asterisk-complete/asterisk/1.8, should I just use the number 10 ? |
| 10:27.09 | ChannelZ | Hmm I just noticed I actually have both versions. I wonder which one it configured to.. |
| 10:27.43 | ChannelZ | well I don't know for sure it would solve anything |
| 10:28.25 | ChannelZ | It's not a hard upgrade but it's kind of throwing a hammer at the problem. Worth a try if you're not afraid |
| 10:28.54 | ChannelZ | It's just strange that you're not getting an outright failure or crash or something, I don't have any guesses as to why it isn't working. |
| 10:29.57 | ChannelZ | briefly browsing the code it seems like it should log errors on most failures so I'm not certain what could actually be going wrong. |
| 10:32.03 | mathi | ChannelZ, I can try to install version 11, but should I install with a package or source code? And how to I actually use 11 instead of 1.8 after installation |
| 10:32.41 | ChannelZ | 11 is beta, best stay away for now |
| 10:32.59 | kannan | anyone has subscribed for Diigum Level 4 support contract ? I am thinking about it. it is USD 8000 p.a |
| 10:33.11 | kannan | *digium* |
| 10:35.37 | ChannelZ | there is an UPGRADE.txt in the source that tells you what all changed between 1.8 and 10 but chances are you're not using anything substantial that changed since you're just starting out |
| 10:35.59 | ChannelZ | just don't do 'make samples' and blow away your existing configs |
| 10:36.46 | ChannelZ | and on that note I need to go to bed |
| 10:36.54 | ChannelZ | good luck |
| 10:37.03 | mathi | thank you:) gn |
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| 12:07.04 | gusto | hey |
| 12:07.15 | gusto | WIMPy: do you have DWissen? |
| 12:07.42 | WIMPy | What's that? |
| 12:07.55 | gusto | radio |
| 12:07.57 | gusto | DLF |
| 12:08.17 | gusto | on DAB+ |
| 12:08.25 | WIMPy | Are they available via IP? |
| 12:08.31 | gusto | of course |
| 12:08.44 | WIMPy | Then I guess I have. |
| 12:08.45 | gusto | http://wissen.dradio.de/ |
| 12:11.10 | WIMPy | Ah, even as vorbis stream. Nice. |
| 12:14.54 | gusto | of course |
| 12:15.01 | gusto | these idiots have everything |
| 12:15.25 | WIMPy | But only for the live stream it seems. That's pretty pointless. |
| 12:15.26 | gusto | and do not see it as something special, every radio i know has a vorbis stream, it seems to be a cost $$$ factor |
| 12:15.40 | gusto | of course |
| 12:15.55 | gusto | all radios have their podcasts/archive in mp3 |
| 12:16.08 | WIMPy | It certainly is, but that doesn;t seem to bother others. |
| 12:16.28 | gusto | like i ve said, they are all idiots |
| 12:17.01 | gusto | the only one radio/tv who have good staff is CRo/CT |
| 12:17.54 | gusto | WIMPy: do you use that speex codec in asterisk? |
| 12:18.34 | WIMPy | No |
| 12:18.45 | gusto | what do you have against that codec? |
| 12:19.10 | WIMPy | Enough bandwidth and no phone that supports it. |
| 12:19.32 | gusto | yes yes |
| 12:19.37 | gusto | that sucks |
| 12:31.39 | mathi | it seems to me that there is a delay between the sound files, e.g. Playback(file1&file2&file3) |
| 12:32.05 | mathi | there will be some default delay when playing, in between the files, am I right ? |
| 12:35.54 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 12:38.16 | gusto | well, and the fs |
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| 12:41.32 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 13:01.52 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 13:58.21 | kannan | any recommendations for hardphones for srtp / tls support as asterisk end points? |
| 14:03.29 | yang | I have some issues with Polycom phone...settings...I am trying to connect it to VOIP uplink |
| 14:03.56 | yang | It doesnt seem to register there, and there are no logs to investigate the issue |
| 14:04.49 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 14:05.06 | yang | hi [TK]D-Fender |
| 14:05.27 | yang | Do you happen to use Polycoms ? |
| 14:06.03 | yang | I think if the line there is registered it comes up with bold letters |
| 14:08.17 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:09.00 | [TK]D-Fender | Polycoms have icons for that |
| 14:09.04 | [TK]D-Fender | not "letters" |
| 14:15.38 | yang | [TK]D-Fender: yes...letters are the ID name |
| 14:15.42 | yang | in front of icon |
| 14:15.44 | *** join/#asterisk Corydon76-home (beige@c-69-137-80-31.hsd1.tn.comcast.net) |
| 14:15.45 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 14:16.00 | [TK]D-Fender | and no, the name doesn't go "bold", the icon changes |
| 14:16.40 | yang | well the other side (my voip unlink) says device unregistered....I think it should register me even if my RTP ports are faulty on my side ? |
| 14:16.49 | yang | voip unlink/uplink |
| 14:17.08 | [TK]D-Fender | registration = SIP, not RTP |
| 14:17.38 | yang | right |
| 14:18.00 | yang | I will make the screenshot of the menus |
| 14:19.18 | [TK]D-Fender | got prove that * is even getting the request |
| 14:19.27 | [TK]D-Fender | go* |
| 14:20.06 | yang | I dont run the very recent firmware but it doesnt matter |
| 14:21.07 | kannan | any recommendations for hardphones for srtp / tls support as asterisk end points? (i have already asked this , sorry to repeat) |
| 14:21.22 | yang | http://v4.si/astscreen/ |
| 14:21.38 | yang | I dunno, maybe values entered are wrong, any hints ? |
| 14:22.06 | [TK]D-Fender | go prove that * is even getting the request <------ |
| 14:23.05 | yang | [TK]D-Fender: oh you know, I dont run a local asterisk...I connect directly to VOIP providers |
| 14:23.11 | yang | with that phone |
| 14:23.56 | yang | so its a bit tricky to analyze |
| 14:24.00 | [TK]D-Fender | What SIP services are you running at that site? |
| 14:24.09 | yang | well |
| 14:24.13 | yang | just their SIP |
| 14:24.28 | [TK]D-Fender | .... on that connection, what else do you have BESIDES the phone? |
| 14:24.29 | yang | they provide login/pasword to connect to their service |
| 14:24.35 | yang | oh |
| 14:24.49 | yang | I have my phone switched into switch, then goes out to the internet |
| 14:25.00 | yang | i defined gateway to match my outgoing gateway |
| 14:25.35 | [TK]D-Fender | What SIP services are you running at that site? <---------------------------- |
| 14:26.18 | yang | Our Internet Phone service is open - that means you can use it with any compatible device. Just use the log in details below to make it work. |
| 14:26.21 | yang | SIP ID:1295424 |
| 14:26.24 | yang | then SIP pass |
| 14:26.26 | yang | just that |
| 14:27.13 | yang | there is an example of phone being tuned in they have screenshot http://www.localphone.com/help/voip/device_guides |
| 14:27.37 | [TK]D-Fender | ... |
| 14:27.52 | yang | I am not sure what you are asking me "hat kind of SIP" |
| 14:28.14 | yang | just SIP service I guess |
| 14:28.36 | [TK]D-Fender | what OTHER VoIP things are you doing over that internet connection that can INTERFERE with the phone? |
| 14:28.46 | yang | none |
| 14:29.06 | [TK]D-Fender | So NO other VoIP devices, servers, etc. Just 1 phone alone? |
| 14:29.48 | yang | oh well I have an unconfigured asterisk session on PC, but I am now terminating it...its not used |
| 14:29.57 | yang | shouldnt interfere |
| 14:30.23 | yang | just 1 phone alone yes |
| 14:30.31 | yang | which goes from NAT to the internet |
| 14:30.47 | [TK]D-Fender | And did you leave any FORWARDING on to that *? |
| 14:30.52 | yang | I just think my polycom settings arent right |
| 14:31.08 | yang | no forwarding rules |
| 14:31.49 | [TK]D-Fender | Go try to register a softphone from there |
| 14:32.35 | yang | why ? It will run on PC then |
| 14:33.16 | yang | i guess I should tcpdump for 5060 |
| 14:33.19 | [TK]D-Fender | facepalms.... |
| 14:33.27 | [TK]D-Fender | Just... not... worth .... it... |
| 14:34.06 | yang | its the wrong values in my polycom phone i am sure about that |
| 14:34.19 | yang | have you looked at screenshots |
| 14:35.41 | yang | actually I might try using local asterisk |
| 14:35.48 | yang | and connect the phone to my asterisk |
| 14:35.52 | yang | and asterisk to my voip uplink |
| 14:36.04 | yang | that way at least i can analyze better |
| 14:37.07 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:41.10 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 14:44.40 | yang | ok asterisk has connected succesfully there |
| 14:50.16 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:50.16 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:54.26 | mathi | hi |
| 14:55.53 | mathi | my IVR canr ead user input with Read(), Background(), ... but how can I record the user? e.g. I want him to leave his name, or a message, and then he would press "#" to continue |
| 14:56.30 | pabelanger | Record() |
| 14:58.24 | mathi | thank you |
| 15:05.26 | yang | [TK]D-Fender: I set it up ok with local asterisk |
| 15:07.02 | yang | Do you happen to know if those +883 numbers are free to call from any mobile operators service ? |
| 15:07.16 | yang | inums ? |
| 15:31.16 | pabelanger | ask your provider, since they are billing you |
| 15:33.06 | *** join/#asterisk YoMomma (~YoMomma@cpe-98-149-120-227.socal.res.rr.com) |
| 15:40.42 | *** join/#asterisk ZX81 (~ZX81@72.189.208.51) |
| 15:41.21 | ZX81 | is svn.digium.com down? |
| 15:42.35 | ZX81 | ~ping |
| 15:42.35 | infobot | ~pong |
| 15:42.48 | ZX81 | ~adn |
| 15:42.48 | infobot | extra, extra, read all about it, adn is hmm... adn is is the Asterisk Daily News - http://www.venturevoip.com/news.php for HTML and http://feeds.feedburner.com/asterisknews for RSS |
| 15:42.54 | ZX81 | heh sweet |
| 15:52.05 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:05.32 | *** join/#asterisk leifmadsen (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:05.32 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:05.39 | leifmadsen | anyone having issues downloading this file? http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8-current.tar.gz |
| 16:11.24 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 16:11.41 | carrar | leifmadsen, ipv6 issues? |
| 16:13.07 | fenrus | cant download it on ipv4 or ipv6 |
| 16:13.29 | fenrus | cant get past HE's network when i trace it |
| 16:13.33 | *** join/#asterisk jsjc (~Adium@213.37.155.15) |
| 16:13.41 | carrar | <PROTECTED> |
| 16:13.44 | carrar | 10 * * * |
| 16:13.47 | carrar | 11 * * * |
| 16:13.58 | fenrus | (ipv6) |
| 16:14.12 | carrar | People still use ipv4? :) |
| 16:14.14 | fenrus | ipv4 i get to; |
| 16:14.14 | fenrus | 26 static-76-164-170-41.apid.com (76.164.170.41) 128.144 ms 127.413 ms 125.944 ms |
| 16:14.17 | fenrus | 27 orc2.api-digital.com (63.238.52.42) 130.351 ms 131.282 ms 131.069 ms |
| 16:16.56 | *** join/#asterisk ZX81 (~ZX81@72.189.208.51) |
| 16:17.54 | carrar | <PROTECTED> |
| 16:17.54 | carrar | 10. 2001:470:7:4c2::2 92.2% 2325. 3099. 2325. 5296. 1465. |
| 16:17.54 | carrar | 11. 2001:470:e0d4::e9 94.1% 2233. 2257. 2233. 2278. 22.5 |
| 16:17.54 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 16:17.56 | carrar | nice |
| 16:18.00 | carrar | 92% packet loss |
| 16:18.09 | carrar | <PROTECTED> |
| 16:18.12 | carrar | 10. downloads.asterisk.org 96% 2 42 7364 5576 6470 7364 |
| 16:18.13 | carrar | same with ipv5 |
| 16:18.14 | carrar | 4 |
| 16:18.26 | carrar | awesome network :) |
| 16:18.47 | fenrus | :D |
| 16:18.51 | fenrus | ipv5, the new black. |
| 16:19.15 | carrar | no uses IPv5 , which means it's less cluttered !! |
| 16:19.16 | fenrus | <PROTECTED> |
| 16:19.20 | carrar | no one |
| 16:19.41 | carrar | more available time slots! |
| 16:20.59 | carrar | leifmadsen, we can sell you internet connectivity :) |
| 16:21.34 | carrar | and SIP trunks too!! :) |
| 16:21.36 | fenrus | i dont work at any provider anymore, but i could probably hook you up with some nice colo in sweden with good connectivity ;) |
| 16:21.53 | carrar | Oh |
| 16:21.54 | carrar | nice |
| 16:22.08 | carrar | Iceberg cooling action? |
| 16:22.12 | fenrus | hehe |
| 16:22.19 | fenrus | and polar bears guarding the place |
| 16:22.26 | fenrus | of course! |
| 16:22.26 | carrar | heh |
| 16:22.52 | carrar | I thought global warming killed off all the polar bears |
| 16:23.15 | fenrus | not quite yet afaik |
| 16:23.37 | fenrus | but i dont really know since we have no polar bears here in sweden ;D |
| 16:25.07 | carrar | Digium needs a datacenter in Seattle, perhaps they just don't know it yet |
| 16:27.38 | tzafrir | ~itsp |
| 16:27.38 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 16:41.57 | *** join/#asterisk aberrios (~aberrios@77.107.183.144) |
| 16:42.05 | aberrios | is packages.asterisk.org down?? |
| 16:52.13 | leifmadsen | aberrios: yes, everything is down right now at the colo |
| 16:52.28 | aberrios | oh bugger... |
| 16:52.34 | aberrios | seems to be back up now... |
| 17:32.06 | *** join/#asterisk Mango45 (~Mango45@d209-89-215-139.abhsia.telus.net) |
| 17:32.28 | Mango45 | Hi. I'm having a curious issue with macros and direct media. |
| 17:32.38 | Mango45 | same => n,Dial(${peer-${EXTEN}},30,M(UsefulInfo)) |
| 17:32.53 | Mango45 | When I use the above, audio goes from the phone to the service provider. |
| 17:32.58 | Mango45 | same => n,Dial(${peer-${EXTEN}},30) |
| 17:33.03 | *** join/#asterisk YoMomma (~YoMomma@cpe-98-149-120-227.socal.res.rr.com) |
| 17:33.08 | Mango45 | With the macro removed, audio goes from the phone directly to the carrier, bypassing the service provider. |
| 17:33.35 | Mango45 | This is the macro: |
| 17:33.50 | Mango45 | exten => s,1,System(/home/mango45/UsefulInfo.sh ${SIPCALLID} &) |
| 17:34.22 | Mango45 | Ideally, I would like audio direct to the carrier, and also run the macro. Does anyone know why it behaves like this? :) |
| 18:06.13 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
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| 18:14.59 | ziz212 | pls let me know few best voip providers that allow sip trunking and sip accounts that can use with asterisk located behind broadband nat ? |
| 18:16.20 | Mango45 | VoIP.ms, Callcentric, Anveo, CallWithUs, |
| 18:16.24 | Mango45 | but you'll never know that. |
| 18:21.22 | [TK]D-Fender | Especially because he wasn't here. |
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| 18:25.06 | Mango45 | If he wishes me to respond, he must insert a pause in there. |
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| 18:37.09 | *** join/#asterisk KingDavidNYC (~IceChat77@pool-74-96-172-86.washdc.fios.verizon.net) |
| 18:37.21 | KingDavidNYC | Hello everyone |
| 18:37.48 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 18:38.15 | KingDavidNYC | can someone please help find out why in the world this Polycom phone I am programming does not appear in the list of peers? Thanks |
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| 23:04.15 | jrgill | Has anyone tried AsteriskWin32? I'm seeking a lightweight implementation of Asterisk as a SIP to PSTN gateway via TAPI-compliant PCI modem. |
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| 23:10.53 | WIMPy | jrgill: You need an Asterisc version 1.0 or older. But you don't really wat that. |
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| 23:22.33 | jrgill | WIMPy, how would version 1.0 or older address my application, as opposed to a current version? |
| 23:22.52 | *** join/#asterisk jsjc (~Adium@213.37.155.15) |
| 23:23.24 | WIMPy | They had chan_modem. |
| 23:25.46 | jrgill | Would a new Asterisk accept an old chan_modem module? |
| 23:26.00 | WIMPy | No |
| 23:26.46 | jrgill | Oh, hmm... |
| 23:27.09 | WIMPy | Get some Linksys SIP gateway. |
| 23:51.17 | CoffeeIV | I am running asterisk 1.8.15.1 |
| 23:53.16 | CoffeeIV | andI have some call quality issues - I have a voip only system, and the outgoing sound is bad but incomign is ok, I have this message when I create a meetme conference: "channel.c:5631 ast_request: No channel type registered for 'DAHDI'" Could that be related to my quality problems ? |
| 23:53.52 | WIMPy | Yes |
| 23:54.00 | WIMPy | You need dahdi for MeetMe. |
| 23:54.16 | WIMPy | Install dahdi or use ConfBridge instead of MeetMe. |
| 23:55.21 | CoffeeIV | I compiled asterisk from source, and did the "./configure --with-dahdi=/usr/src/dahdi-2.2.1+dfsg-1ubuntu2/" so I think I should have dahdi installed. lsmod shows the dahdi kernel module installed. |
| 23:56.07 | WIMPy | The should be autodetected. |
| 23:56.33 | WIMPy | Then make sure res_timing_dahdi is loaded. |
| 23:56.48 | WIMPy | From the message it seems you also need chan_dahdi loaded. |
| 23:57.02 | CoffeeIV | ok, I had forgotten about res_timing_dahdi, let me check that |
| 23:57.14 | WIMPy | You might have to provide a valid configuration file for it to load. |