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| 13:05.03 | webphage | i have a problem where a client will disconnect from MeetMe conference without BYE (network issues) then attempt to rejoin conference. number of conference participants increases each time, and client audio stutters/is jittery. |
| 13:05.36 | webphage | was told session timers would help, but i have a new question now: won't client still refresh both sessions after rejoining conference? |
| 13:19.57 | webphage | i assumed that client was still receiving two audio streams which causes the stutter, so if it's still receiving both streams won't it refresh both timers as well? |
| 13:21.22 | webphage | i am using asterisk 1.6.0.26, and peers-0.4.2 (minimal java SIP stack) |
| 13:25.27 | webphage | actually i'm not even sure yet if my asterisk version supports session-timers, but my question still stands. |
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| 13:57.24 | blackslikz | hello people ... i use ael ... but i am kind new i want to add music on hold to my dialer so that it can wait as long as i want then call it back with an hash key... no idea how to do it |
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| 14:04.22 | sekil | webphage: no I don't think client will refresh both sessions |
| 14:04.59 | sekil | webphage: sst will make asterisk hangup the call if the reinvites are not acked |
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| 14:22.16 | blackslikz | hello people ... i use ael ... but i am kind new i want to add music on hold to my dialer so that it can wait as long as i want then call it back with an hash key... no idea how to do it |
| 14:30.19 | webphage | sekil: that's what I'm wondering. I guess this is a SIP question, how is it that only one reinvite will be acked? |
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| 14:38.21 | sekil | not only one |
| 14:38.38 | webphage | sekil: i mean if a reinvite is sent via both channels |
| 14:38.45 | webphage | sekil: two reinvites |
| 14:38.54 | webphage | sekil: won't the client ack both? |
| 14:39.15 | webphage | sekil: OH |
| 14:39.21 | sekil | if the line is/was down no |
| 14:39.22 | webphage | sekil: I think I understand. |
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| 14:39.47 | sekil | the points is...the reinvites will come at any given period ..say 60 secs.. |
| 14:40.01 | sekil | if they're not 200ok-ed...l |
| 14:40.11 | sekil | then asterisk bye's the call |
| 14:40.36 | webphage | sekil: I was confused between the audio stream and the open channels, re-INVITEs are sent via channel, and only one is actually active |
| 14:40.44 | webphage | sekil: so now I understand, thank you! |
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| 18:38.27 | hesco1 | My /root/build_asterisk.sh script dated October 2012 left behind on my now legacy pbx server used libss7 and new_prompts as files I wget'd and built into that system. I'm not seeing those on the site now. Are they now obsolete? |
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| 19:20.15 | *** join/#asterisk infobot (ibot@rikers.org) |
| 19:20.15 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.12.0 (2014/08/19), 1.8.30.0 (2014/08/19); Standard: Asterisk 12.5.0 (2014/08/19); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 20:42.48 | WIMPy | hesco1: I don't know what newprompts is or was, but you need libss7 if you need ss7. |
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| 21:03.04 | hesco1 | WIMPy: what is ss7 and how do I know if I need it; and if I need it, where would I find it? |
| 21:03.21 | hesco1 | are the voice prompts now a part of the base asterisk project? |
| 21:04.04 | Penguin | If you don't know what it is, you probably don't need it. |
| 21:06.08 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 21:08.06 | hesco1 | ok, reading about it on voip-info leaves me thinking that my sip providers likely need it for call termination, but that I do not need it to communicate with the sip providers. |
| 21:08.25 | Penguin | ~itsp |
| 21:08.25 | 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 $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 21:08.28 | hesco1 | or with the sip phones I host |
| 21:08.33 | *** join/#asterisk cmendes0101| (~cmendes01@pool-98-112-76-22.lsanca.fios.verizon.net) |
| 21:10.03 | hesco1 | ok, thanks Penguin. can you enlighten me as to whether those prompts which were a separate package two years ago are now included in asterisk core? |
| 21:10.40 | Penguin | If you told me a file name or the words in the prompt, I could look it up. |
| 21:12.16 | WIMPy | doesn;t know any changes in the last two years. |
| 21:16.15 | Penguin | Or you can grab the file and check it yourself. http://downloads.asterisk.org/pub/telephony/sounds/asterisk-core-sounds-en-wav-current.tar.gz |
| 21:16.45 | Penguin | If you don't find your prompts in core sounds, check extra sounds. http://downloads.asterisk.org/pub/telephony/sounds/asterisk-extra-sounds-en-wav-current.tar.gz |
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| 22:15.54 | *** join/#asterisk wonderworld (~ww@ip-62-143-157-238.hsi01.unitymediagroup.de) |
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| 22:26.34 | *** join/#asterisk Milenco (~Milenco@home.milenco.net) |
| 22:27.22 | Milenco | Hey guys! Hoping you can help me with this conditional set which is giving me an error |
| 22:27.33 | Milenco | same => 6,Set(_FALLBACK_VOIP=${IF($[${TRANSPORT}="SIP/voicetrading"]?"SIP/flowroute":"SIP/voicetrading")}) |
| 22:27.54 | Milenco | give me this warning: WARNING[22391]: ast_expr2.y:1488 op_div: non-numeric argument |
| 22:28.14 | Milenco | I can't figure out what I'm doing wrong and cant find working examples :( |
| 22:43.41 | *** join/#asterisk fury_ (~fury@znc.hq.codexterous.com) |
| 22:45.40 | Penguin | You have lots of misplaced quote marks. |
| 22:46.55 | Penguin | Set(_FALLBACK_VOIP=${IF($[${TRANSPORT}=SIP/voicetrading]?SIP/flowroute:SIP/voicetrading)}) |
| 22:49.59 | fury_ | hello! I've been given a username, password, phone number, and proxy IP from my SIP provider, and I'm trying to configure asterisk - I have the following simple sip.conf so far: http://www.codexterous.com/paste/results/rDceuD89.html |
| 22:50.02 | fury_ | does that look right? |
| 22:50.28 | fury_ | I'm sure there are other problems with it, but the main one is this: [Sep 6 18:42:30] WARNING[-1]: chan_sip.c:21424 handle_response_register: Got 404 Not found on SIP register to service cyberlead837@69.60.226.14, giving up |
| 22:50.28 | fury_ | <PROTECTED> |
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| 22:54.27 | *** join/#asterisk P-NuT (~P-NuT@2.216.84.81) |
| 22:55.04 | Penguin | fury_: No, there are several problems with that. |
| 22:55.43 | fury_ | great. such as what? I'm sure there are many, but I'd like to nail down the 404 thing first |
| 22:55.49 | Penguin | fury_: Is your asterisk behind a NAT router? |
| 22:56.08 | Milenco | Thanks Penguin! :) |
| 22:56.13 | fury_ | it actually *is* the NAT router, and has both public and private interfaces |
| 22:56.15 | Milenco | I found it in that syntax on several sites |
| 22:56.32 | Milenco | I did fix it already using Gotoif (little less nice) |
| 22:56.39 | Milenco | I'll try your solution later on! |
| 22:56.43 | Penguin | Consider ExecIf() as well. |
| 22:57.42 | P-NuT | Hey all. |
| 22:57.42 | P-NuT | Does anyone have any good tutorial for using SIP SIMPLE in asterisk 11? |
| 22:58.48 | Milenco | Yeah I saw it later on :) |
| 23:00.11 | Penguin | fury_: What is this cyberlead837@69.60.226.14 thing you've mentioned? That isn't a VoIP.ms username nor their IP address. |
| 23:00.56 | fury_ | I'm not using voip.ms, but I did copy that configuration from elsewhere |
| 23:01.07 | fury_ | it's actually vianet.ca (have a friend who works there) |
| 23:01.14 | fury_ | he's not around currently though |
| 23:01.45 | Penguin | milenco: The combination of ExecIf(Set()) or Set(myVar=${IF()}) depends on a couple things. "Do I need to set a variable, but I could set it one of two ways... or is my need to set the variable contingent on something else?" |
| 23:02.33 | Penguin | fury_: It's not very intuitive to name your peer "voipms" and ask for help with it, without ever saying you're not even working on voip.ms at all. |
| 23:02.52 | Milenco | What I wanted to do it set a fallback provider (to Voicetrading by default unless Voicetrading is already my primary line) |
| 23:03.06 | fury_ | I didn't think it mattered what provider I'm using, also I didn't even know that was the provider name. |
| 23:03.15 | fury_ | Nevertheless, I've changed the name. |
| 23:04.18 | Milenco | I see you only removed my quotation marks Penguin, when _should_ I use them when defining variables? |
| 23:04.40 | Milenco | Because GotoIf does accept them |
| 23:04.45 | Penguin | fury_: Let's cover some stuff that is in the book... If your asterisk is behind a NAT, you set the externaddr value to your public IP address, set nat=yes, and directmedia=no. |
| 23:05.11 | Penguin | milenco: Use quotation marks when you want them to be part of your string. |
| 23:05.17 | Penguin | They are literal. |
| 23:05.59 | fury_ | I can either use a NAT or not use a NAT, as this *is* the NAT. So public would be fine, I just have to make sure the firewall rules are set up. since that's probably easier, I can do it that way. externip=mydomain.com, nat=no and directmedia=yes then? |
| 23:06.09 | Penguin | milenco: Set(myVar="value") does not mean myVar will equal value. It will equal "value". |
| 23:06.28 | Milenco | ahh |
| 23:06.39 | Penguin | externip expects a number. |
| 23:06.45 | Milenco | so I use quotes for setting vars and no quotes for evaluating them |
| 23:06.58 | Penguin | Nope. |
| 23:07.01 | Penguin | Not what I said. |
| 23:07.11 | Penguin | Because value does not equal "value". |
| 23:07.27 | Penguin | They are not the same. Quotes are literal and are part of your string. |
| 23:07.32 | Penguin | value = value |
| 23:07.38 | Penguin | "value" = "value" |
| 23:07.43 | Penguin | "value = "value |
| 23:07.50 | Penguin | va"lue = va"lue |
| 23:07.54 | Milenco | ahh, i think I understand |
| 23:08.17 | Milenco | i could use some string with quotes as long as I do it consistently for that string |
| 23:08.33 | Milenco | thanks again Penguin :> |
| 23:09.02 | Penguin | When writing conditions, such as in a GotoIf(), if you don't quote your variable reference, don't expect it to match when you quote the test value. |
| 23:09.41 | Penguin | GotoIf($[${myVar} = "value"]?...) |
| 23:10.08 | Penguin | This will not match if the variable's content does not also have quotes in it. |
| 23:10.39 | Penguin | Also, empty variables in tests like that will cause problems. |
| 23:10.53 | Penguin | You can quote the variable but you must also quote the test value. |
| 23:11.05 | Penguin | Or use another character to make it non-null. |
| 23:11.18 | Penguin | GotoIf($[x${myVar} = xvalue]?...) |
| 23:11.38 | Penguin | If myVar is null, you will be comparing x to xvalue. |
| 23:14.18 | Penguin | If you quote it, you'd be comparing "" to "xvalue" and those quotes are parts of the string, not ignored. |
| 23:14.26 | Penguin | sorry, "value" not "xvalue" |
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| 23:45.56 | *** join/#asterisk matrix1233 (~matrix123@41.231.9.186) |
| 23:46.37 | matrix1233 | hi |
| 23:47.21 | matrix1233 | how can i do a transfert to a extention ? i have tryed on zoiper but i can't |
| 23:49.12 | matrix1233 | ?? |
| 23:53.19 | *** join/#asterisk gnudna (~gnudna@unaffiliated/sklav) |
| 23:54.41 | gnudna | hi guys can a rule like this be added to inbound context exten => 4162221111,1,Hangup() no asterisk just drops the call aka does not answer it |
| 23:55.34 | gnudna | i rather not pay to answer scam calls and then hangup on them |