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01:40.46 | jmmills | can someone explain to me why a sip provider would be sending (or asterisk be receiving) a plus symbol in an incoming calls extension? |
01:41.50 | p3nguin | Some weird providers think that's how it is supposed to be done. |
01:42.29 | p3nguin | You have two choices. Ask them to change it, or accept it as the extension they send calls to (change your dialplan). |
01:43.13 | jmmills | is there a way I can just strip the leading + early in the context? |
01:44.25 | p3nguin | I wouldn't bother. Just change your inbound extension. |
01:44.50 | p3nguin | In such a condition, it's going to be purely cosmetic. |
01:44.58 | p3nguin | Now in a caller ID number, it's annoying for a few reasons. |
01:47.33 | Gugge | in callerid i just change + to 00 |
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01:54.16 | FreezingCold | http://en.alldaytalk.com/Cell/ < Can I do something like this with a Asterisk box? |
02:06.39 | p3nguin | freezingcold: Absolutely. |
02:07.34 | p3nguin | It's very simple to do either type. |
02:07.54 | FreezingCold | I want to do the first |
02:08.12 | p3nguin | Do you have a cell phone plan with fave 5, my 5, my circle? |
02:08.31 | FreezingCold | Yeah, I can get one number unlimited for $7/m |
02:08.33 | FreezingCold | I've very new to Asterisk, any guides? |
02:08.39 | FreezingCold | Or do you know SIPSorcery? |
02:08.39 | p3nguin | ~book |
02:08.39 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
02:09.28 | p3nguin | Here's what I would do: get a free DID with sipgate or ipcomms, add that number to your free airtime on your cell plan. |
02:10.01 | p3nguin | Are you wanting to make your outbound calls over a landline at home, or using an ITSP? |
02:10.07 | FreezingCold | Yeah, I already kinda have that. I have a freephoneline account (with SIP settings) |
02:10.09 | FreezingCold | I'm in Canada |
02:10.18 | FreezingCold | On my cell phone |
02:10.27 | Maliuta | FreezingCold: where in canadia? |
02:10.31 | FreezingCold | Ottawa |
02:10.35 | FreezingCold | Using Telus |
02:10.50 | Maliuta | FreezingCold: My parents live in Fort MacMurray alberta |
02:12.08 | FreezingCold | Never been there |
02:12.10 | p3nguin | You'll make free calls to your Asterisk system from your cell phone... then your Asterisk system will make phone calls out to someone else at a lower cost per minute than your airtime. |
02:12.17 | FreezingCold | Yeah, that's the idea |
02:12.24 | FreezingCold | Now how would I go about doing that? |
02:13.16 | Maliuta | FreezingCold: is there a reason you don't just SIP from your phone directly into the * system? |
02:13.50 | FreezingCold | Because I don't want to use up data on my cell phone |
02:14.50 | p3nguin | (2110.00) <p3nguin> Are you wanting to make your outbound calls over a landline at home, or using an ITSP? <--- still waiting for the answer for this. |
02:15.02 | FreezingCold | ITSP |
02:15.13 | p3nguin | Do you have one with low per-minute rates already? |
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02:15.28 | FreezingCold | I get unlimited in Canada & USA for free |
02:15.29 | FreezingCold | so yeah |
02:15.37 | FreezingCold | way cheaper then 20 cents a minute |
02:15.59 | p3nguin | Alright, so all you have to do is configure your DID onto your Asterisk system... |
02:16.18 | p3nguin | Devise a simple authenticate/DISA dialplan. |
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02:16.28 | FreezingCold | Which isn't simple for someone like me =P |
02:17.04 | FreezingCold | Any premade type stuff? |
02:17.05 | p3nguin | I can do it for you. Give me a minute to write an example. |
02:17.09 | FreezingCold | =0 |
02:17.10 | FreezingCold | Thanks! |
02:17.44 | FreezingCold | On a totally different topic, should I put my ATA in DMZ? |
02:18.01 | FreezingCold | I've been having issues with one way audio sometimes |
02:18.25 | p3nguin | no |
02:18.31 | FreezingCold | Why not? |
02:18.34 | FreezingCold | What's wrong with DMZ? |
02:18.38 | p3nguin | You should never use DMZ. Configure the port forwarding properly. |
02:18.44 | FreezingCold | already did |
02:18.50 | FreezingCold | still have the issue |
02:19.06 | FreezingCold | http://sipsorcery.wordpress.com/2009/08/05/nat-rtp-and-audio-problems/ |
02:19.33 | FreezingCold | That's what's happening to me |
02:26.27 | p3nguin | freezingcold: http://pastebin.com/LUSCAdY1 |
02:26.55 | p3nguin | freezingcold: Let me explain it. |
02:27.27 | FreezingCold | alright |
02:27.55 | p3nguin | freezingcold: This dialplan says that your DID number is 8005551212 (put your actual DID here). If someone calls it, it will hangup, unless the callerID matches 3149691077 (use your cell phone number here). |
02:28.14 | FreezingCold | go on |
02:28.50 | p3nguin | freezingcold: If the caller ID matches, then it will run the Authenticate() command, which asks you to enter a PIN. In my example, the PIN is store in the ast database (you can type the actual PIN in the dialplan if you want). |
02:29.07 | p3nguin | freezingcold: If you provide the right PIN, it will give you a dial tone. |
02:29.07 | FreezingCold | got it |
02:29.34 | p3nguin | Crap, I just realized I missed the underscores on the pattern matching for the magic context. |
02:29.37 | FreezingCold | I doubt you know anything about it, but could I port that plan over to SIPSorcery? |
02:29.51 | p3nguin | http://pastebin.com/MtF4rKU7 corrected |
02:29.54 | FreezingCold | That way I don't need to leave my PC on 24/7 |
02:30.18 | p3nguin | (I have a tendency to forget my underscores.) |
02:31.18 | p3nguin | When you get that dialtone, you dial the number. DISA() sends the number into the [magic] context, where your Asterisk system dials out using a SIP peer by the name of your-itsp-peer in this example. |
02:31.55 | p3nguin | That's the basic way to duplicate Plan A - Magic Number. |
02:32.03 | FreezingCold | Nice |
02:32.20 | FreezingCold | How could I take incoming calls as well? |
02:32.35 | p3nguin | I probably wouldn't ever give anyone your magic number, which you added to your free airtime. |
02:32.52 | p3nguin | If you want to use it for dual-purpose, you just change the dialplan. |
02:33.16 | p3nguin | Rather than running Hangup() when someone else calls, branch into something more useful... which does not provide them the dial tone. |
02:33.25 | FreezingCold | yeah |
02:35.30 | p3nguin | It's pretty easy to change that behavior. Instead of the hangup, you could use a Goto() to send them into a useful context. |
02:35.45 | FreezingCold | Have you used SIPSorcery before? |
02:36.25 | p3nguin | Nope. |
02:37.03 | FreezingCold | Hmm, guess I'm on my own for porting it |
02:40.52 | p3nguin | freezingcold: Here's a variation for allowing other people to call in without having the ability to call back out: http://pastebin.com/X5xetHvC |
02:42.18 | p3nguin | You _could_ technically use this concept to set up something similar for friends/family giving everyone his own PIN and dial-out rules. |
02:44.46 | Gibby | p3nguin, got it working thanks, can I add a 2nd D() option to dial a 3rd option? |
02:45.35 | p3nguin | freezingcold: You could also do away with the Authenticate() command and use DISA's own PIN system from a file which determines the context based on the PIN entered. |
02:45.50 | p3nguin | gibby: I actually don't know. I never tried that. |
02:46.13 | Gibby | ok, i will let you know thanks |
02:52.43 | Gibby | what is the package for the CLI? i can issue asterisk -r and it connects but no other commands work then |
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03:12.10 | p3nguin | gibby: Sounds broken. |
03:12.43 | Gibby | p3nguin, that is what i thought, trying an update |
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03:33.34 | Gibby | i am troubleshooting why my inbound calls are not coming through, i have forwarded 5060 and 10000-20000, i do a tcp dump on my router of my external IP and I see 5060 come through but nothing else, that seems to mean an issue with my SIP provider right? |
03:34.02 | [TK]D-Fender | Gibby: What does "see 5060 come through" mean? |
03:34.37 | [TK]D-Fender | Gibby: And just forwarding ports isn't enough |
03:34.57 | Gibby | i can see the initial 5060 UDP packets come in from the internet from IP of my SIP provider and the correctly get forwarded to asterisk server but I do not see any other packets even attempt to come in |
03:36.14 | [TK]D-Fender | Gibby: And who says the packets you see come in are meaningful? |
03:36.40 | [TK]D-Fender | Gibby: You also fail to describe what other packets you are expecting, and why. |
03:37.32 | Gibby | from what I was reading the 5060 packets are kinda of the initiator but the UDP channel traffic is carried over ports between 10000 and 20000 |
03:38.31 | [TK]D-Fender | Gibby: Where do I see that there is even a call coming in? Or that * is trying to answer it? |
03:38.46 | [TK]D-Fender | Gibby: Of that audio is supposed to be flowing? |
03:40.33 | Gibby | I am doing a TCP dump on the interface of my external traffic on my router, I set the source or destionation as the IP of my SIP provider, during an outbound call I see some UDP 5060 packets then packets on ports between 10000 and 20000, when i try to make an incoming call, i only see the UDP 5060 packets come in and I get a busy signal |
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03:51.25 | [TK]D-Fender | Gibby: Maybe, just MAYBE you should be actually looking at the content of those packets in real detail. |
03:51.44 | [TK]D-Fender | Gibby: 5060 is CALL CONTROL. This could include a telling GTFO messagin in there somewhere. |
03:51.47 | Gibby | zuj, i will check them |
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04:17.02 | Gibby | [TK]D-Fender, do you want to look at these packets? |
04:17.48 | [TK]D-Fender | Gibby: Do you want an opinion on them? |
04:17.55 | Gibby | that would be great |
04:20.57 | Gibby | trying to send via IRC |
04:20.59 | [TK]D-Fender | Gibby: PASTEBIN |
04:21.01 | [TK]D-Fender | !pb |
04:21.03 | [TK]D-Fender | ~pb |
04:21.03 | 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.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. |
04:22.49 | Gibby | http://pastebin.com/EKFaQP20 |
04:23.47 | [TK]D-Fender | Gibby: Trash. Dow go get * SIP DEBUG from * CLI |
04:23.50 | [TK]D-Fender | Now* |
04:24.24 | p3nguin | What the heck is that from? |
04:24.47 | Gibby | tcpdump of raw data packets, easily viewed in Wireshark |
04:26.30 | [TK]D-Fender | Gibby: Don't care about wireshark. No go get a full capture of the attempt from * CLI with SIP DEBUG enabled |
04:27.58 | Gibby | getting it now |
04:30.06 | Gibby | looks like NAT issue, thanks guys |
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05:08.52 | Gibby | internet dropped, don't know if you got my last msg but, i see it NAT'ing now, still seeing SIP/2.0 401 Unauthroized though |
05:09.36 | p3nguin | Authorize it. |
05:10.23 | [TK]D-Fender | No. Over half an hour later and still no pastebin as requested. To late for help on my side. |
05:10.27 | [TK]D-Fender | heads off to bed |
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05:15.58 | Gibby | Authorizing it would be in the sip.conf right? |
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05:17.50 | p3nguin | If the problem is that you don't have a peer matching what is sending the call, yes. |
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05:19.37 | Gibby | ok |
05:20.07 | Gibby | that is the type= in sip.conf? |
05:22.03 | Gibby | looks like options of type= are peer user and friend |
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05:36.19 | Gibby | fyi got it working, took everything out of my router that I had configured for Asterisk port forwarding and sip, setup sip_nat.conf and added the following to my sip.conf, changed type to friend and added nat=yes, qualify=yes, insecure=very, dtmfmode=auto, dtmf=inbound, disallow=all and allow-ulaw&gsm |
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05:55.27 | ssh-add | good morning |
06:03.16 | raden | how can i tell asterisk to only process call files from 9 am to 6 pm ? |
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06:27.22 | kaldemar | raden: set the file creation time between 9 am and 6 pm before moving it to the spool directory. |
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06:39.36 | ssh-add | raden you can also do it in the dialplan |
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07:01.23 | verywiseman | what are family and key operators which are found in DB(family/key) function? |
07:07.30 | kaldemar | verywiseman: what you want them to be. |
07:09.21 | kaldemar | the database has families with key-value pairs in them. by adding stuff you define the family and the key names at the same time. |
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07:40.24 | hurdman | hi |
07:41.35 | ssh-add | hello |
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08:07.34 | hrhrhr | is 1.6 feature frozen now? all updates being bug fixes? |
08:13.20 | wdoekes2 | I've been told it is |
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08:33.55 | hrhrhr | changelog doesn't make for easy reading does it |
08:34.04 | hrhrhr | 10 merged with svnfeed |
08:34.05 | hrhrhr | goto 10 |
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08:45.45 | nunners | Sorry to ask a freepbx question, but it's a little but urgent... does anyone know of a problem when setting up ring groups on freepbx 2.8.0.3 with the dialparties.cgi script? I keep getting utils.c ast_carefulwrite: write() returned error: Broken pipe |
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09:07.36 | SeTTleR | hi |
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10:29.43 | eMBee | good evening |
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10:55.46 | rayno_b | Hi there everyone, I need some assistance. My system is not showing the number where an incoming call comes from. I have no clue how to correct this. |
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11:03.22 | kaldemar | rayno_b: how and where do you expect it to show the number? |
11:04.23 | rayno_b | kaldemar -> when the call is received (in CLI) it comes in from ''. I'm expecting that DAHDI is not getting the number correctly from the Telco. Eventually, I'd like to see the number where a call comes from on the SIP phones receiving the calls. |
11:05.18 | kaldemar | what technology are you using the interface telco? analog? PRI? BRI? |
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11:06.56 | rayno_b | BRI (Digium B410p Card) |
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11:09.35 | kaldemar | see parameters usecallerid, cidsignalling and cidstart in chan_dahdi.conf.sample. "pri intense debug span X" (unless the command has changed in recent versions) will show you if there is a number to show in the signalling. |
11:13.18 | WIMPy | pri set debug 2 span X |
11:13.54 | rayno_b | is this on the CLI? |
11:14.00 | kaldemar | rayno_b: yes |
11:16.20 | rayno_b | I get: No such command 'pri set debug 2 span X' (type 'help pri set' for other possible commands) |
11:16.58 | kaldemar | replace X with your span number |
11:16.59 | WIMPy | Then it's the old version: pri intense debug span X |
11:17.18 | WIMPy | Err. Yes |
11:17.47 | kaldemar | pri show ... will get you the right span number. use tab completion. |
11:18.31 | rayno_b | was: pri intensive debug span 1 |
11:18.35 | rayno_b | let me phone in |
11:20.37 | rayno_b | I grepped the log file now for that number, but I don't think it's coming in with the number. Could DAHDI be configured wrongly? |
11:21.36 | WIMPy | If you don't get it, you don't get it. |
11:23.16 | kaldemar | that's what the telco is sending you. |
11:26.48 | SeTTleR | this should be stored in ${CALLERID(num)} correct? maybe you can explicitly print that to the console with the Verbose() app |
11:27.02 | SeTTleR | from your dialplan |
11:27.30 | SeTTleR | exten => s,n,Verbose(call came from ${CALLERID(num)}) |
11:27.34 | SeTTleR | for example |
11:27.56 | SeTTleR | then you know, you don't get a callerid |
11:28.11 | WIMPy | If there's nothing in the debug, it won't get into any variable. |
11:28.27 | SeTTleR | :-) |
11:28.27 | SeTTleR | that's true |
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11:37.04 | rayno_b | I just want to make 100% sure - It can't maybe mean that there's just something wrong with DAHDI config? |
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12:01.08 | SeTTleR | is there a way to define custom log levels in asterisk? i want to create log entries from the dialplan with Log() and they should end up in separate files. of course, i can separate them with syslog, but maybe this can be done easily with asterisk |
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12:33.05 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0-rc3 (2010/10/07), 1.6.2.13 (2010/09/15), 1.4.36 (2010/09/15), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.4 (2010/09/01) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs,#astricon |
12:41.21 | *** join/#asterisk Maximo (~maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
12:44.16 | *** join/#asterisk nicola_pav (~chatzilla@mail2.tikalnetworks.com) |
12:44.26 | *** join/#asterisk BANSAL (~bansal@117.199.112.4) |
12:45.47 | nicola_pav | hello. I have asterisk servers. I connected two ip phones. I can make call from one another. everything is fine. upon running tcpdump on asterisk i get icmp unreachable errors |
12:46.08 | nicola_pav | tunning wireshark it seems icmp port unreachable code 3 |
12:46.15 | nicola_pav | how can i solve this? |
12:48.25 | [TK]D-Fender | nicola_pav: What are you doing with wireshark and it sounds like the call is WORKING, so what is the actual problem? |
12:50.18 | nicola_pav | [TK]D-Fender: the call is working yes |
12:50.26 | nicola_pav | but why i am getting those icmp messages? |
12:50.38 | nicola_pav | i want a normal call flow without any errors |
12:50.57 | SeTTleR | ? where do you get those messages? in wireshark? who cares? |
12:51.03 | [TK]D-Fender | nicola_pav: If your call is fine then you simply don't know how to use wireshark or don't know what you're supopsed to be looking for with it |
12:51.15 | nicola_pav | i run tcpdump and every 5 seconds i am having them |
12:51.23 | nicola_pav | i just analyzed it more with wireshark |
12:51.26 | SeTTleR | maybe you have icmp blocked by a firewall or turned of somewhere in the proc/ fs |
12:51.55 | nicola_pav | is it normal while in the call flow to have every 5 seconds unreachable icmp errors? |
12:52.10 | nicola_pav | i dont want those errors to appear |
12:52.27 | SeTTleR | maybe you have icmp blocked by a firewall or turned of somewhere in the proc/ fs <-- |
12:52.47 | nicola_pav | what do u suggest? |
12:52.55 | SeTTleR | i really don't think that this is an asterisk issue |
12:52.58 | nicola_pav | what shall i do with proc/? |
12:53.08 | nicola_pav | is it ip phone issue? |
12:53.39 | [TK]D-Fender | nicola_pav: MAYBE YOU SHOULD show us |
12:54.23 | nicola_pav | of course |
12:54.30 | SeTTleR | what does cat /proc/sys/net/ipv4/icmp_echo_ignore_all say? |
12:54.36 | SeTTleR | on both machines |
12:54.53 | nicola_pav | i can run it on asterisk |
12:54.56 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
12:54.57 | *** mode/#asterisk [+o malcolmd] by ChanServ |
12:54.58 | nicola_pav | the other is ip phone |
12:54.58 | SeTTleR | if one says 1, then you have the problem |
12:55.10 | SeTTleR | ok, try that |
12:55.14 | nicola_pav | on asterisk |
12:55.16 | nicola_pav | its 0 |
12:55.29 | nicola_pav | 14:39:14.396368 IP 192.168.199.11 > 192.168.199.201: ICMP 192.168.199.11 udp port 11794 unreachable, length 208 |
12:55.35 | nicola_pav | this message is from tcpdump |
12:55.50 | nicola_pav | 192.168.199.11 is the IP address of the ip phone |
12:56.14 | nicola_pav | i get such messages each 5 seconds |
12:56.32 | SeTTleR | huh.. i don't know whireshark that well, but why is there something with udp? |
12:56.52 | nicola_pav | this message is from tcpdump |
12:57.09 | nicola_pav | i captured another call with wireshark to analayze more |
12:57.17 | nicola_pav | it seems CODE 3 |
12:57.23 | nicola_pav | which means port unreachable |
12:57.37 | SeTTleR | well this is not icmp not reachable |
12:58.01 | nicola_pav | it is |
12:58.08 | nicola_pav | icmp unreacable and code 3 |
12:58.11 | SeTTleR | i don't know code 3 or something.. the error message above says something about udp?! |
12:58.14 | nicola_pav | which means port unreachable |
12:59.06 | nicola_pav | i got it from wireshark |
12:59.24 | nicola_pav | maybe i am wrong |
12:59.31 | nicola_pav | i need to stop those messages |
12:59.49 | SeTTleR | i think the problem is another one.. the asterisk tries to reach the ip phone with udp port 11794 and this is closed on the phone which returns with the icmp message from above |
12:59.50 | [TK]D-Fender | nicDo you? You haven't told us a PROBLEM this is related to. |
13:00.27 | nicola_pav | i am afraid i did understand u |
13:00.42 | nicola_pav | y not stopping those messages? |
13:00.48 | nicola_pav | they appear each 5 seconds |
13:01.07 | SeTTleR | can be some kind of misconfiguration |
13:01.20 | SeTTleR | rtp ports or something.. i don't know |
13:01.37 | SeTTleR | asterisk tries to reach phone on that port.. port is closed -> message appears |
13:01.46 | nicola_pav | any hint? |
13:02.22 | SeTTleR | check the rtp ports in sip.conf and on the phone |
13:03.32 | nicola_pav | on the phone there is this parameter |
13:03.40 | nicola_pav | udp keep alive-message |
13:03.47 | nicola_pav | does it play a role? |
13:03.54 | SeTTleR | maybe |
13:04.56 | SeTTleR | i don't know what phone you are using, but this sounds like "i am sending a udp message periodically every 5 seconds and trigger a message in wireshark with that" :D |
13:04.57 | nicola_pav | its disabled |
13:05.03 | nicola_pav | i will enable it and see |
13:05.29 | nicola_pav | because it happens each 5 seconds |
13:05.42 | nicola_pav | u say that sth is sending periodically? |
13:05.44 | nicola_pav | right? |
13:05.53 | [TK]D-Fender | qualify=yse <- |
13:05.55 | [TK]D-Fender | yse |
13:06.34 | nicola_pav | qualify=yes for the account? |
13:06.37 | nicola_pav | i will check now |
13:08.34 | nicola_pav | i have qualify yes already |
13:09.19 | [TK]D-Fender | nicola_pav: turn it OFF and see if it goes away |
13:09.28 | nicola_pav | ok |
13:10.55 | nicola_pav | same |
13:11.02 | nicola_pav | turned it off and same |
13:11.05 | nicola_pav | each 5 seconds |
13:11.34 | [TK]D-Fender | nicola_pav: Aside fromt he fact you can see that message in wireshark, is there an actual PROBLEM? |
13:12.02 | nicola_pav | are those messages normal? |
13:12.13 | nicola_pav | for me they r not |
13:12.15 | [TK]D-Fender | nicola_pav: Maybe. Is there a PROBLEM? |
13:12.16 | *** join/#asterisk Woody2143 (~Woody2143@machine76.Level3.com) |
13:12.26 | nicola_pav | depends :) |
13:12.35 | nicola_pav | for me. there is a problem |
13:13.50 | [TK]D-Fender | writes nicola_pav off as the problem and moves on to more productive matters |
13:14.50 | nicola_pav | y can't anyone help me to figure it out? |
13:15.50 | *** join/#asterisk coppice (~chatzilla@112.203.17.210.dyn.pacific.net.hk) |
13:16.02 | [TK]D-Fender | nicola_pav: So far you are looking at the RESULT, and not the CAUSE |
13:16.39 | *** join/#asterisk UQlev (~Yuriy@212.50.99.8) |
13:16.54 | nicola_pav | can anyone direct me to the cause? |
13:19.17 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
13:20.35 | SeTTleR | well this is simple network debugging. look, what packet causes the phone to send the icmp message, observe with netstat, what process tries to send that packet and there you are.. |
13:21.43 | nicola_pav | what options should i run with netstat? |
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13:26.47 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
13:27.53 | raden | how do i set the maximum amount of calls being placed at once for call files ? |
13:28.01 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
13:30.02 | [TK]D-Fender | raden: No such thing. Stop placing them all there at once yoursle |
13:30.04 | [TK]D-Fender | f |
13:30.48 | raden | does each call need to be a seperate file ? |
13:31.50 | raden | so I should be basically having PHP write a file and then cron script move 1 file every minute to asterisk spool ? |
13:32.12 | hurdman | i'm looking for a sphinx 4 integration into asterisk, i have tried some tuto (and patch) found in the web, but it seems to be out-of-date, anyone have a recent link ? |
13:32.22 | Katty | HERRO WORLD |
13:33.03 | [TK]D-Fender | raden: Create wahtever scheduling mechanism you feel like but if there are multiple in there they will ALL egt processed immediately |
13:33.17 | [TK]D-Fender | Katty: Mew. |
13:33.42 | raden | [TK]D-Fender, so if i put 4,000 calls in there they will all start calliing ? |
13:33.50 | raden | Morning Katty :) |
13:34.14 | [TK]D-Fender | raden: Yes |
13:34.40 | [TK]D-Fender | raden: And that many simultaneous might be a nasty load for your server and it could have a heart attack |
13:34.54 | raden | totally understandable |
13:36.23 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
13:36.35 | hrhrhr | is there a way to remove the local card from the asterisk server for pri connections |
13:36.44 | Katty | hi raden |
13:36.46 | Katty | hugs raden |
13:36.48 | hrhrhr | i.e. use some external device for interfacing with it |
13:36.48 | Katty | hugs [TK]D-Fender |
13:36.59 | Katty | would anyone like half a headache? |
13:37.04 | hrhrhr | hi fives Katty |
13:37.12 | Katty | hrhrhr: Hello. |
13:37.18 | [TK]D-Fender | hrhrhr: For interfacing with the PRI... or the CARD? |
13:37.39 | hrhrhr | interfacing with the pri |
13:37.43 | hrhrhr | an external card, if you like |
13:37.44 | hrhrhr | or something |
13:39.06 | [TK]D-Fender | hrhrhr: There are tons of SIP gateways, etc. All very pricey by comparison |
13:39.39 | hrhrhr | i want to retain the pri local interface |
13:39.44 | hrhrhr | but |
13:39.52 | hrhrhr | i dont want to have to open a machine if the pbx breaks |
13:40.01 | hrhrhr | nor do i want to buy 2 * pri per site |
13:40.08 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
13:40.12 | hrhrhr | 2 * pri card rather |
13:40.42 | WIMPy | hrhrhr: And what do you do if the gateway breaks? |
13:40.47 | [TK]D-Fender | hrhrhr: Why have you jsut introduced multiple SITES into this picture? Please clarify your situation |
13:41.29 | hrhrhr | how does one build a hot/warm standby pbx |
13:41.35 | hrhrhr | when the interface card lives in the machine |
13:41.51 | hrhrhr | of course, if the interface card/device dies, that's something else |
13:41.57 | hrhrhr | but i want to cover machine/server failure |
13:42.22 | [TK]D-Fender | hrhrhr: Then get a PRI>SIP interface |
13:42.40 | [TK]D-Fender | hrhrhr: eg AudioCodes Mediant series |
13:43.46 | hrhrhr | ok cheers |
13:48.45 | creativx | man I am going crazy trying to figure out what is chopping up our SIP audio |
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13:57.21 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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14:04.24 | Katty | Naikrovek: poke |
14:05.17 | Naikrovek | sup |
14:05.37 | Naikrovek | sorry lot of stuff going on today |
14:05.48 | Naikrovek | stressful stuff |
14:07.11 | *** join/#asterisk Marquel (~Marquel@84.16.240.149) |
14:07.18 | Marquel | morning. |
14:07.32 | Marquel | it seems my asterisk has a problem using time-triggered includes. |
14:09.02 | Marquel | syntax for asterisk 1.6 is 'include => newcontext|08:00-10:00|mon-wed|*|*' to include newcontext every monday, tuesday and wednesday from 08:00 to 10:00, right? |
14:09.06 | Katty | Naikrovek: i saw your fb thing. was just curious if you were doign ok |
14:12.20 | *** join/#asterisk b0gatyr_ (~b0gatyr_@adsl-146-67-241.mia.bellsouth.net) |
14:15.36 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
14:15.37 | *** join/#asterisk xoveruk (~xover@193.220.59.2) |
14:15.41 | xoveruk | hi all |
14:18.02 | *** join/#asterisk theHub (~karl@69.177.93.21) |
14:19.13 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
14:20.12 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
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14:22.21 | xoveruk | If calls are conneting but then dropping after 20-30secs, what would that suggest is the issue or where should I start looking or clues? |
14:23.01 | mechbangirc | hi one of my client says he has a shared ip so he can not forward port 5060 to an internal asterisk server. i do not know much about networking and the whole blame is on me |
14:24.06 | xoveruk | mechbangirc: how can he have a shared IP, does he mean dynamic? |
14:25.06 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
14:25.31 | mechbangirc | xoveruk: i doubt his knowldge and skill. i told him if he has dynamic ip its still can be done. though he keep saying it is not possible i dont have static ip my ip is sharred!!! i dont understand |
14:25.58 | xoveruk | he has a DHCP address from his IP. |
14:26.05 | Naikrovek | Katty: not really |
14:26.06 | Naikrovek | thanks though |
14:26.32 | xoveruk | So he needs to configure DDNS on his internet router and then use that DNS address for configuring the trunks or SIP accounts. |
14:26.50 | Naikrovek | people find what they can do to stress me out, then they do that one single thing only, over and over and over and over and over and over and over and over |
14:27.11 | Naikrovek | after i tell them to stop, after i tell them what else to do, after i tell them that it's stressing me out |
14:27.12 | Naikrovek | over |
14:27.13 | Naikrovek | and over |
14:27.17 | Naikrovek | and i'm tired of it |
14:27.27 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
14:27.36 | WIMPy | Naikrovek: Isn't it nice if you can count on people? *eg* |
14:27.56 | Naikrovek | the longer i live the more i see that you can't count on anyone for anything at all |
14:28.02 | Naikrovek | even the simplest things |
14:28.07 | xoveruk | Naikrovek: I 2nd that |
14:28.14 | Naikrovek | i even have to tell my wife what to EAT half the time |
14:28.17 | Naikrovek | can't you feed yourself |
14:28.29 | Naikrovek | can FEEL his hair greying |
14:28.48 | xoveruk | mechbangirc: Do you understand what I am suggesting? |
14:28.50 | WIMPy | You still got hair? |
14:29.50 | leifmadsen | Naikrovek: amen :) |
14:29.57 | leifmadsen | Naikrovek: become a recluse -- it's easier |
14:30.07 | mechbangirc | xoveruk: sorry i was away. internal server is on static ip (172.16.1.2) or so. |
14:31.17 | p3nguin | mechbangirc: I assume it's a shared server with only one IP address. |
14:31.23 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
14:31.41 | drmessano | mechbangirc, are they using a wireless ISP? |
14:31.52 | mechbangirc | xoveruk: on wan side his ip is dynamic i dont understand sharred ip at all. |
14:32.01 | mechbangirc | drmessano: no it is wired |
14:32.23 | drmessano | Who is the provider? |
14:33.51 | mechbangirc | drmessano: hey i know for a fact the provider provides dsl connection not cable or something so how can it be sharred. |
14:33.54 | p3nguin | mechbangirc: (0931.17) <p3nguin> mechbangirc: I assume it's a shared server with only one IP address. |
14:34.40 | mechbangirc | p3nguin: no it is a dedicated asterisk server with one private address |
14:35.48 | xoveruk | how can you have a shared IP, doesnt make any sense, how can you route internet traffic |
14:35.55 | drmessano | mechbangirc, just guessing, maybe he's got a single port DSL modem/router, and has a router behind it.. |
14:36.23 | drmessano | Just trying to guess what the client is talking about.. who knows how much he really knows |
14:36.35 | xoveruk | can someone tell me what the valid comments forms for asterisk config files. ';' and '#'? |
14:36.51 | p3nguin | mechbangirc: Then it's not a shared IP address. |
14:37.06 | p3nguin | mechbangirc: Tell him to learn some networking terminology before saying he can't comply with your request. |
14:38.03 | mechbangirc | p3nguin: i guess i have to tell him. its hard to tell |
14:38.18 | schmidts | xoveruk its ; |
14:38.24 | p3nguin | http://www.100best-free-web-space.com/articles/paid-web-hosting/shared-or-dedicated-ip-addresses-44.html |
14:38.57 | p3nguin | shared means SHARED, or not dedicated. |
14:39.15 | p3nguin | Dynamic and shared are mutually exclusive terms. |
14:39.50 | p3nguin | You could have a shared DYNAMIC address, or you could have a shared STATIC address. |
14:40.09 | p3nguin | Similarly, you could have a dedicated dynamic address, or you could have a dedicated static address. |
14:40.10 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
14:40.25 | xoveruk | schmidts: what does the '#' denote/ |
14:40.53 | leifmadsen | xoveruk: DAHDI related files use # for comments |
14:40.56 | leifmadsen | the rest use ; |
14:41.09 | leifmadsen | it's a historical thing |
14:41.50 | mechbangirc | p3nguin: i think shared add is good for web hosting companies not for small companies who do not provide public services over internet |
14:41.51 | xoveruk | @leifmadsen: so both are valid comment files, the later referring to relating files? |
14:42.04 | leifmadsen | xoveruk: they are both valid in the same file |
14:42.10 | leifmadsen | xoveruk: oops wrong |
14:42.21 | leifmadsen | xoveruk: I meant, they are not both valid in separate files, but not the same file |
14:42.40 | mechbangirc | p3nguin: i would throw this stuff on him |
14:42.43 | leifmadsen | xoveruk: some files use ; while others use #. The ones that use # are pretty much all DAHDI related (i.e. chan_dahdi.conf) |
14:43.04 | drmessano | So DAHDI likes hash? |
14:43.04 | leifmadsen | xoveruk: you can tell which file uses what by looking at the sample files |
14:43.12 | leifmadsen | drmessano: who doesn't? |
14:43.18 | drmessano | Good point |
14:43.29 | leifmadsen | drmessano: people I don't want to hang out with, that's who! |
14:43.38 | schmidts | :d |
14:43.44 | drmessano | lol |
14:43.59 | *** join/#asterisk S-chum32 (~S-chum32@68.188.5.142) |
14:44.06 | leifmadsen | ok time to get focused and get some work done before lunch |
14:45.04 | drmessano | There has to be an additional joke in there about DADHI, hash, and a semi-colon |
14:45.21 | p3nguin | mechbangirc: There's also an exclusivity between public and private addressing. |
14:45.21 | drmessano | My head is too stopped up to wrap around it |
14:45.52 | hrhrhr | what are your thoughts on tdmoe as opposed to a pri<>sip gateway |
14:46.05 | WIMPy | drmessano: Head already on dahdi-land? |
14:46.34 | WIMPy | in |
14:46.37 | *** join/#asterisk [cannibalera] (~cannibale@201-41-239-137.fnsce703.dsl.brasiltelecom.net.br) |
14:46.43 | p3nguin | s/addressing./addressing, and shared and dedicated./ |
14:46.55 | SeTTleR | pigpen, in case you like to know: the problem with system() yesterday was fixed. was my fault, as the sh binary was not in the chroot. confusing error messages from asterisk though... but thanks for your help :D |
14:47.36 | schmidts | hrhrhr direct connection needed which can be hard if your customer wants a fallback system on different positions like min. 5 miles away ;) |
14:47.54 | *** join/#asterisk mechbangirc (~mechbangi@mbl-65-157-110.dsl.net.pk) |
14:48.42 | pigpen | SeTTleR, sure...glad to see you found it. |
14:49.09 | SeTTleR | me too :D |
14:49.10 | WIMPy | Jo, forget about those ideas of using vi as login shell. Asterisk as login shell rulez! |
14:50.44 | drmessano | WIMPy, my head feels like someone is crushing it with Asterisk 1.2, I have Zaptel running out my nostrils, and I am running to the restroom every 5 minutes to take a trixbox |
14:52.56 | *** part/#asterisk [cannibalera] (~cannibale@201-41-239-137.fnsce703.dsl.brasiltelecom.net.br) |
14:53.05 | kaldemar | prescribe yourself 1.8 mg's of asterisk and some DAHDI. |
14:53.29 | SeTTleR | <PROTECTED> |
14:58.09 | drmessano | lol |
14:59.40 | Naikrovek | leifmadsen: you're no recluse. |
15:00.20 | leifmadsen | Naikrovek: shut your mouth |
15:00.28 | thehar | nom nom donut! |
15:00.33 | Naikrovek | ... |
15:00.37 | leifmadsen | thehar: nom nom is so 2007 |
15:00.48 | thehar | eat my **** |
15:00.52 | thehar | ****sucker |
15:00.54 | thehar | yeap |
15:01.04 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
15:01.14 | leifmadsen | and then we pink bellied The_Boy_Wonder! |
15:01.20 | leifmadsen | ohai! |
15:01.24 | thehar | lol |
15:02.09 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
15:03.17 | p3nguin | leifmadsen: So what's the new method of devouring yummy foods? |
15:03.18 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
15:03.23 | leifmadsen | p3nguin: chewing? |
15:03.29 | thehar | chewing is overrated |
15:03.36 | leifmadsen | I personally just thing nomnomnom is retarded in a bad way |
15:03.42 | leifmadsen | s/thing/think |
15:03.52 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
15:04.11 | drmessano | Yeah, nom nom sounds like you're ...... |
15:04.15 | p3nguin | Is anything ever retarded in a good way? |
15:04.22 | drmessano | AYBABTU was |
15:04.36 | thehar | p3nguin: yes leif |
15:04.39 | drmessano | I think even O RLY made sense |
15:04.41 | thehar | comma that somewhere |
15:04.51 | leifmadsen | O'Reilly?! O RLY! |
15:05.01 | drmessano | O RRY was my fav |
15:05.01 | p3nguin | orally |
15:05.30 | drmessano | By the time we got to OMGWTFBBQ, I knew O RLY was over |
15:05.40 | leifmadsen | lol |
15:05.53 | thehar | lol |
15:09.29 | Katty | dear universe. |
15:09.35 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net) |
15:09.45 | Katty | please stop tortuing me. my stomach hurts, that's enough now. can we please make up and be friends now? |
15:09.48 | Katty | love, katty |
15:09.57 | thehar | gets his magical wand |
15:10.07 | thehar | uses it on Katty |
15:10.08 | thehar | fixed. |
15:10.13 | Katty | horay! |
15:10.35 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
15:10.40 | *** part/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net) |
15:13.49 | Marquel | nobody? |
15:15.02 | thehar | for fun, I thought I'd try registering someschmuck.com, but some schmuck already owns it. |
15:17.45 | p3nguin | irony |
15:17.49 | thehar | heh |
15:27.09 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net) |
15:27.12 | *** join/#asterisk TobSnyder (~schneider@dslb-088-073-198-122.pools.arcor-ip.net) |
15:28.11 | fullstop | can emailsubject and emailbody be set for an individual voicemail entry? I have some vm contexts where I specify emailbody= and emailsubject=, but they are never used. |
15:28.48 | fullstop | I added some debug code to app_voicemail.c, and I don't see them being parsed.. |
15:29.13 | [TK]D-Fender | fullstop: parms aren't context based |
15:30.28 | fullstop | [TK]D-Fender: one moment |
15:30.50 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
15:31.27 | fullstop | [TK]D-Fender: http://svn.digium.com/svn/asterisk/branches/1.6.2/configs/voicemail.conf.sample |
15:31.38 | fullstop | search for Bianca |
15:32.04 | [TK]D-Fender | fullstop: That's news.... |
15:32.30 | [TK]D-Fender | fullstop: seems to imply it is possible |
15:32.51 | p3nguin | fullstop: If you're using a version that supports it, why wouldn't it be possible? |
15:33.09 | fullstop | I'm asking because it is not working for me. :) |
15:33.56 | fullstop | I see where it looks for emailsubject and emailbody in the [global] |
15:34.16 | fullstop | and, I dump everything that it parses.. but it appears to only get that information from users.conf and not voicemail.conf |
15:35.38 | [TK]D-Fender | fullstop: Sorry.. users.conf voids your warranty :p |
15:35.52 | fullstop | [TK]D-Fender: nothing in it except for [global] |
15:35.55 | fullstop | let me remove it |
15:36.23 | [TK]D-Fender | fullstop: http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.conf <--- seems to contradidict those options as being alloed at the box level anyway |
15:37.37 | fullstop | [TK]D-Fender: the patch mentioned appears as if it was applied to 1.6.x |
15:38.17 | *** join/#asterisk Nwab (~Schnitzel@unaffiliated/benwa) |
15:38.20 | [TK]D-Fender | fullstop: What patch? |
15:38.25 | p3nguin | specifically, the 1.6.2 branch. |
15:38.27 | fullstop | https://issues.asterisk.org/view.php?id=14372 |
15:39.18 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
15:39.34 | [TK]D-Fender | fullstop: that patch implies REALTIME. Are you using realtime? |
15:39.43 | fullstop | negative |
15:39.55 | rrb3942 | anyone familiar with issues between dialogic and asterisk, that can causes the rtp for the audio periodically just start/stop in a call? |
15:40.07 | [TK]D-Fender | fullstop: then it sounds like a wish-list item that isn't real |
15:40.24 | fullstop | [TK]D-Fender: Are you sure that you are not German? |
15:40.36 | [TK]D-Fender | rrb3942: Dialogic doesn't speak RTP last I checked..... |
15:40.45 | [TK]D-Fender | fullstop: Entirely. |
15:40.45 | rrb3942 | on a SIP call |
15:40.56 | fullstop | [TK]D-Fender: I don't see anything within the patch code which is realtime specific. Let me look a bit further. |
15:41.05 | rrb3942 | i thought they had some media gateway boxes that supported SIP? |
15:43.46 | coppice | [TK]D-Fender: its not the 20th century any more. most of dialogic's products speak RTP |
15:47.18 | [TK]D-Fender | coppice: And he didn't specify what product he is dealing with. |
15:47.37 | [TK]D-Fender | coppice: So 20th century is entirely possible |
15:47.39 | rrb3942 | wish I knew, it is someone elses dialogic box |
15:47.48 | [TK]D-Fender | WOW... such awesome details. |
15:48.04 | [TK]D-Fender | Hi, my Ford doesn't work... HOW TO FIX PLAESE!?!? |
15:50.55 | rrb3942 | hey, I know the details suck, I've been trying to get information from these guys for two days now and its just circles with them |
15:51.04 | drmessano | I can understand why |
15:51.25 | coppice | [TK]D-Fender trade it for a Honda |
15:53.14 | neurosys | or a personality :P |
15:54.44 | [TK]D-Fender | neurosys: Putting the FUNF back in DYSFUNCTIONAL |
15:54.47 | [TK]D-Fender | FUN* |
15:54.49 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
15:54.54 | neurosys | hehe |
15:57.20 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
15:59.53 | xoveruk | how can I tell which port is being used for SIP? |
16:00.18 | madduck | e.g. netatst -naup (5060/udp) |
16:00.24 | madduck | netstat even |
16:00.48 | *** join/#asterisk _Sam-- (~sam@unaffiliated/sam--/x-573746) |
16:01.13 | _Sam-- | i realize this is the asterisk channel, however, im wondering if anyone can give a thumbs up to the switchvox hardware devices? |
16:02.13 | russellb | gives a thumbs up |
16:02.17 | *** join/#asterisk f00bar80 (~eng.debia@41.239.209.51) |
16:02.19 | russellb | also works for digium ;-) |
16:02.34 | f00bar80 | Just i'm asking can i have the vps hosting serving both web hosting and IP-pbx hosting if i'm gonna to install asterisk + freepbx and cPanel/whm ??? |
16:02.36 | russellb | _Sam--: you can also try #switchvox |
16:02.40 | _Sam-- | thank you. |
16:05.06 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
16:05.09 | wcselby | o/ |
16:05.17 | wcselby | anyone here use queuemetrics? |
16:06.21 | *** join/#asterisk drudge` (tacos@unaffiliated/drudge/x-837452) |
16:07.50 | fullstop | [TK]D-Fender: okay, I dug through the VM code a bit more. emailsubject and emailbody are only applied if they are present in users.conf. |
16:08.11 | fullstop | Considering something so evil, a lot of asterisk revolves around it. :P |
16:10.07 | *** join/#asterisk razu (~razu@razu.data.ee) |
16:11.43 | xoveruk | madduck: thanks |
16:11.56 | xoveruk | I also used nmap and found the firewall settings, where can I set the port number for SIP? |
16:14.10 | ManxPower | xoveruk, the sip port in your firewall? |
16:14.49 | ManxPower | fullstop, you must be using some kind of GUI |
16:16.03 | fullstop | ManxPower: No, I'm not. |
16:16.18 | fullstop | ManxPower: look at the code. |
16:16.30 | ManxPower | fullstop, emailsubject and emailbody do not, as far as I know, depend on users.conf |
16:16.39 | fullstop | ManxPower: one moment |
16:17.03 | fullstop | ManxPower: http://pastebin.com/ZihXzvq5 |
16:17.04 | ManxPower | if there is a users.conf what you see might be the case. |
16:17.11 | xoveruk | ManxPower: no for asterisk |
16:17.12 | ManxPower | But only crazy people use users.conf |
16:17.18 | fullstop | I am not using users.conf |
16:17.27 | fullstop | ucfg represents users.conf |
16:17.29 | fullstop | and.. |
16:17.37 | ManxPower | xoveruk, if you do that none of your sip clients are likely to work anymore |
16:17.45 | fullstop | appy_options_full is only called if you are using users.conf or asterisk realtime. |
16:17.54 | xoveruk | ManxPower: For each sip device, the port numbered listed, what is that for? Connecting to the device? |
16:18.04 | fullstop | apply_options_full is what reads emailsubject and emailbody. |
16:18.13 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
16:18.31 | ManxPower | xoveruk, remember ALL IP has TWO ports. the SOURCE port and the DESTINATION port. The SOURCE port is chosen dynamically by the OS, the DEST port can't change or nothing will connect to it. |
16:18.50 | ManxPower | fullstop, then file it as a bug, because it is one. |
16:19.13 | ManxPower | the ports listed in sip show peers would be the source port |
16:20.26 | xoveruk | source port meaning? the device that is connecting to asterisk? |
16:20.36 | *** join/#asterisk rtert (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net) |
16:20.41 | rtert | hi |
16:20.44 | rtert | is there anyone here ? |
16:20.48 | thehar | nope. |
16:20.50 | thehar | sleeping |
16:21.00 | rtert | ok leave it |
16:21.59 | ManxPower | xoveruk, the source of the connection, usually the device |
16:22.21 | rtert | lol |
16:22.25 | rtert | hi there |
16:22.28 | ManxPower | ~ask |
16:22.29 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
16:22.40 | xoveruk | ManxPower: ok great stuff |
16:23.03 | rtert | i don't if it the write chnnel, i need an asterisk expert for a project |
16:23.26 | rtert | is there any channel for that ? |
16:24.23 | ManxPower | you can try #asterisk-biz if you are looking to hire a consultant |
16:26.32 | rtert | ok thanks a lot |
16:27.12 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
16:29.23 | *** join/#asterisk jdwjr_ (~jon@66.195.223.170) |
16:36.14 | *** join/#asterisk salviadud (~ralfalfa@189.224.63.254) |
16:50.46 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
16:57.49 | *** join/#asterisk b_d (~brian@mail.bsg.bm) |
17:03.01 | *** join/#asterisk c0rnoTa (~c0rnoTa@80.251.113.51) |
17:03.30 | *** part/#asterisk c0rnoTa (~c0rnoTa@80.251.113.51) |
17:05.49 | *** join/#asterisk zplinux (~zplinux@213.8.57.217) |
17:06.15 | zplinux | can I get some help with the asterisk rpm spec file? |
17:06.43 | zplinux | in line 110 I read: %{?_without_alsa:%if 0} what does that mean? |
17:07.57 | Katty | peeks in |
17:08.10 | Katty | ohai |
17:10.08 | Katty | zplinux: how're you today? |
17:10.25 | zplinux | fine dear, and how are you? |
17:10.34 | Katty | oh just peachy, thanks. |
17:11.16 | zplinux | say, where can I get help with the asterisk rpm spec file? |
17:11.43 | Katty | i would imagine someone here would be able to help you eventually. |
17:11.53 | Katty | you will have to be patient tho. |
17:12.00 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
17:14.10 | Katty | Naikrovek: we need to talk :< |
17:14.49 | Naikrovek | not really in the mood right now, katty |
17:15.03 | Naikrovek | gotta run cable through the office today |
17:15.06 | Katty | k, well i'll take a raincheck. |
17:15.06 | thehar | ew |
17:15.34 | f00bar80 | any comment ? |
17:15.35 | f00bar80 | Just i'm asking can i have the vps hosting serving both web hosting and IP-pbx hosting if i'm gonna to install asterisk + freepbx and cPanel/whm ??? |
17:16.08 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
17:16.34 | thehar | are you going to do SIP signalling and RTP traffic on different ports while running both? |
17:16.48 | *** join/#asterisk csnook (~chris@138.210.3.1) |
17:21.26 | TobSnyder | what is relaxdtmf for? |
17:21.48 | [TK]D-Fender | TobSnyder: Exactly what it says. Relaxing the DTMF detection routine |
17:24.08 | Katty | i'll relax your dtmf in a minute. |
17:24.15 | thehar | i'll do it in 30 seconds |
17:24.51 | Katty | that's quick. |
17:25.13 | Katty | perhaps even premature. |
17:25.45 | [TK]D-Fender | That's what SHE said |
17:25.46 | coppice | anything below 36 weeks is considered premature |
17:25.52 | thehar | lol |
17:26.13 | Katty | that sounds painful. |
17:26.26 | TobSnyder | [TK]D-Fender: any more detail? use cases? |
17:27.08 | [TK]D-Fender | TobSnyder: Getting unreliable DTMF? Try it |
17:27.24 | TobSnyder | just have problems with DTMF on FXS Port |
17:27.34 | drmessano | Get a new phone |
17:27.48 | drmessano | Take that one back to Walmart and exchange it |
17:28.10 | [TK]D-Fender | TobSnyder: go try the option then. |
17:28.10 | TobSnyder | seems not to be related with phone, as it occurs on all analog telephone sets I have tested so far |
17:28.11 | Katty | walmart :< |
17:28.43 | TobSnyder | and just wondered in dahdi show channel XX what "Relax DTMF: no" stands for |
17:28.43 | Katty | i had some dtmf problems once, but it ended up somehow being motherboard related. |
17:28.47 | *** part/#asterisk rtert (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net) |
17:28.49 | carrar | You'd think walmart would have more walls in their store |
17:29.20 | Katty | yes. they are the mart of wal. |
17:29.37 | TobSnyder | first I thought it might be a problem of double detection (as astribank an also detect dtmf) but this seems not to be the reason |
17:29.38 | Katty | still not a fan tho. i feel like white trash the minute i park. |
17:29.48 | TobSnyder | so I am now checking relaxdtmf |
17:29.48 | Katty | TobSnyder: does it happen with a soft phone? |
17:29.59 | drmessano | Walmart was named afer it's two founders, Walt Disney and Martin Short |
17:30.01 | carrar | Katty, you have to wear sweatpants there |
17:30.05 | TobSnyder | Katty: as far as I have tested not |
17:30.05 | drmessano | Everyone knows that |
17:30.06 | carrar | and a white teeshir |
17:30.07 | carrar | t |
17:30.08 | Katty | i have sweat pants. |
17:30.16 | Katty | but they say PINK on the back |
17:30.18 | TobSnyder | Katty: x-lite seemed to have no problems with dtmf |
17:30.30 | carrar | you have to wear them to walmart |
17:30.33 | drmessano | Try with a different phone |
17:30.42 | Katty | drmessano: he's tried with several hardware phones. |
17:30.52 | Katty | TobSnyder: what if you send digits with the dial plan? |
17:30.54 | drmessano | Get a new FXS card |
17:31.02 | Katty | TobSnyder: and then have it playback them to you |
17:31.05 | drmessano | Maybe you are listening too hard |
17:31.11 | Katty | TobSnyder: does it also have issues |
17:31.58 | carrar | I need to install my TV Antenna today |
17:32.00 | Katty | TobSnyder: what lvl is the dtmf getting snickerdoodled up? |
17:32.09 | Katty | carrar: wanna set up a fta box for me? |
17:32.19 | carrar | FTA? |
17:32.23 | Katty | free to air |
17:32.26 | carrar | oh |
17:32.33 | carrar | SURE |
17:32.35 | carrar | Fly my out |
17:32.38 | Katty | :>>> k |
17:32.39 | carrar | me |
17:32.44 | carrar | I can install anything |
17:32.46 | Katty | makes arrangements. |
17:32.55 | carrar | I have crimpers too |
17:32.55 | [TK]D-Fender | [13:30]<TobSnyder>Katty: x-lite seemed to have no problems with dtmf <- because it isn't AUDIO |
17:33.14 | carrar | and several free multimeters from harbor frieght :) |
17:33.17 | Katty | carrar: crimpers? |
17:33.26 | carrar | to put ends on the RG6 |
17:33.29 | Katty | oh |
17:33.32 | thehar | Katty: i still need to make snickerdooles |
17:33.32 | Katty | i was thinking like.... |
17:33.38 | Katty | you're going to do my hair?! |
17:33.43 | carrar | YES |
17:33.48 | Katty | no. |
17:33.54 | Katty | oh. |
17:33.55 | carrar | I might use RJ45 crimpers for that |
17:33.58 | Katty | are you metro? |
17:34.01 | carrar | hell no |
17:34.04 | Katty | if you are metro, you can do my hair. |
17:34.16 | Katty | then keep your crimpers away from me. |
17:34.19 | carrar | I live in the metro downtown city for some unknown reason however |
17:34.27 | Katty | i am sorry. |
17:34.43 | carrar | I can see 30+ Access Points from my place |
17:35.02 | Katty | Eesha. i can only see 4 |
17:35.08 | Katty | But I also have trees. |
17:35.15 | carrar | I have trees |
17:35.19 | carrar | 7 trees |
17:35.21 | *** join/#asterisk Letoric (~Letoric@tpi-dfw-uc-f1-69-94-238-22.totalprocess.net) |
17:35.21 | thehar | lol |
17:35.24 | Katty | :< |
17:35.24 | TobSnyder | Katty: TobSnyder: what lvl is the dtmf getting snickerdoodled up? ?????? |
17:35.39 | carrar | 4 of them are 40+ feet |
17:35.48 | carrar | at least |
17:35.48 | Katty | Oh. I forgot not everyone speaks Kat. |
17:36.18 | TobSnyder | actually when sending DTMF directly via Dialplan (e.g. as parameter of Dial Command) it worked |
17:36.19 | Katty | TobSnyder: from phone to asterisk box, from the fxs card (see log)...played back from the telco? i'd start swapping around hardware. |
17:36.39 | Katty | TobSnyder: again, that is not audio. |
17:36.40 | TobSnyder | I have created an extension that is playing back digits |
17:36.54 | Katty | TobSnyder: same with a software phone. that's not audio either. |
17:37.24 | TobSnyder | [dtmf-test-b] |
17:37.24 | TobSnyder | exten => 972,1,Dial(DAHDI/r1/01803001179,180,D(w0w7w1w6w4w1)) |
17:37.48 | TobSnyder | attention, 3 lines are pasted: |
17:37.50 | TobSnyder | [dtmf-test-a] |
17:37.50 | TobSnyder | exten => 971,1,Answer |
17:37.50 | TobSnyder | exten => 971,2,Goto(saydigits,s,1) |
17:37.55 | Katty | carrar: those aren't real trees :< |
17:38.10 | carrar | sure they are |
17:38.13 | carrar | they are evergreens |
17:38.15 | Katty | :< |
17:38.17 | Katty | :<<< |
17:38.29 | carrar | might be 60' |
17:38.33 | carrar | who knows |
17:40.34 | [TK]D-Fender | TobSnyder: Your description is crap. Where do I see the end that is FXS in there? Where do I see a failed call? Where do we see what the GOTO leads to? |
17:40.45 | *** join/#asterisk angryuser_laptop (~Greg@90-156-167-83.reverse.alphalink.fr) |
17:40.47 | Katty | carrar: http://www.flickr.com/photos/midmophil/2968628604/sizes/l/in/photostream/ |
17:40.55 | TobSnyder | sry |
17:41.23 | carrar | yeah those are big |
17:41.26 | angryuser_laptop | Good day, when dahdi_cfg show FXS > its FXS or FXO ( it was inverted as i remember) |
17:41.29 | Katty | trees = <3 |
17:42.09 | tzafrir | angryuser_laptop: look at the output of lsdahdi |
17:42.21 | tzafrir | the first column should normally get it right |
17:42.24 | Katty | carrar: also http://www.flickr.com/photos/whitebuffalobk/3371824508/sizes/l/in/photostream/ <- |
17:43.18 | carrar | Where is the photo of you hugging that tree? |
17:43.18 | beardy | Hi Katty |
17:43.53 | Katty | hi beardy |
17:43.56 | Katty | carrar: there isn't one. |
17:44.46 | TobSnyder | so hwo to debug? |
17:45.09 | carrar | switch to sip phones |
17:45.18 | [TK]D-Fender | TobSnyder: What debug? Where's the failed CALL? Where is the REST of your code? PASETBIN <--------- |
17:45.19 | carrar | and call it good |
17:46.09 | beardy | Katty: I'm about to have pizza, want some? |
17:46.22 | carrar | Epizza? |
17:46.29 | angryuser_laptop | tzafrir, thanks |
17:46.52 | TobSnyder | [TK]D-Fender: I don't have failed calls, the problem is that some IVR don't recognize DTMF correctly when calling from analog telephone set. When looking at logs you won't see anything interesting related to that problem there, as the call is directly bridged between FXS and BRI Port when it is established |
17:47.03 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
17:47.19 | [TK]D-Fender | TobSnyder: Show us this etst you are comparing |
17:47.22 | [TK]D-Fender | test* |
17:47.43 | angryuser_laptop | TobSnyder, try to set relaxdtmf=yes |
17:48.04 | [TK]D-Fender | anyWe already told him to try it |
17:48.11 | [TK]D-Fender | angryuser ^ |
17:48.16 | angryuser_laptop | good |
17:48.19 | Katty | beardy: mmmm, what kind? |
17:48.26 | [TK]D-Fender | angryuser_laptop: And he was the one who brought it up actually |
17:48.39 | angryuser_laptop | [TK]D-Fender, very good |
17:49.23 | angryuser_laptop | try to make the dtmf duration bigger, and check if your regional settions is ok |
17:49.28 | TobSnyder | http://pastebin.com/2Dtu0upu |
17:49.53 | *** join/#asterisk Tim_Toady (~moi@193.92.224.201.dsl.dyn.forthnet.gr) |
17:50.32 | TobSnyder | let me just check relaxdtms |
17:50.35 | TobSnyder | *f |
17:51.25 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
17:53.05 | [TK]D-Fender | TobSnyder: We told you to enable it half an hour agi |
17:53.08 | [TK]D-Fender | ago* |
17:53.17 | TobSnyder | well but problem is that some digits are doubled |
17:53.41 | TobSnyder | just thought relaxdtmf will result in more digits bying recognized |
17:54.26 | [TK]D-Fender | TobSnyder: Show us your test attempt |
17:56.14 | TobSnyder | [TK]D-Fender: you want to see full logs? |
17:56.19 | beardy | Katty: mincemeat, tomato, onion, spice tomato and garlic sauce, cheese :) |
17:56.38 | beardy | spicy* |
17:57.29 | TobSnyder | [TK]D-Fender: http://pastebin.com/zhzDRXrG |
17:58.31 | TobSnyder | relaxdtmf did not change anything |
17:58.41 | TobSnyder | concerning the issue |
18:00.20 | TobSnyder | argh |
18:01.24 | *** join/#asterisk nort22 (~nort@208.104.101.168) |
18:02.08 | TobSnyder | after restarting everythin I now encountered that relaxdtmf seems to solve the issue, I am going to do some more tests |
18:02.25 | [TK]D-Fender | ....... |
18:02.30 | TobSnyder | sorry I'm kinda confused |
18:02.31 | [TK]D-Fender | has wasted his time again |
18:03.07 | TobSnyder | [TK]D-Fender; perhaps I can spend you some beer some day |
18:03.40 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
18:04.57 | TobSnyder | ok still have problems |
18:05.14 | TobSnyder | I hate those problems "sometimes it works, sometimes not" ....... |
18:07.21 | *** join/#asterisk DelphiWorld (~Delphi@41.200.13.124) |
18:07.25 | zplinux | I still seek help with the asterisk spec file |
18:07.36 | zplinux | what does %{?_without_tds:%if 0} mean? |
18:07.59 | *** part/#asterisk DelphiWorld (~Delphi@41.200.13.124) |
18:08.49 | Qwell | zplinux: magic |
18:09.28 | *** join/#asterisk DelphiWorld (~Delphi@41.200.13.124) |
18:09.35 | DelphiWorld | [TK]D-Fender: tu parle français? |
18:09.59 | [TK]D-Fender | DelphiWorld: Certainment |
18:10.16 | DelphiWorld | [TK]D-Fender: tré bien, je sais pas sa! je suis que vous êtes un canadiain |
18:10.21 | fullstop | leifmadsen: the patch I just submitted, you changed it to "needs license". Is there something I need to do? |
18:10.29 | DelphiWorld | [TK]D-Fender: tré bien, je sais pas sa! je suis sure que vous êtes un canadiain |
18:10.35 | [TK]D-Fender | DelphiWorld: Il te manque un verbe la-dans.... |
18:10.36 | leifmadsen | fullstop: no, just wait for the license to be approved |
18:10.43 | fullstop | leifmadsen: thanks |
18:10.48 | DelphiWorld | [TK]D-Fender: oui, j'ai just oublié |
18:11.36 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
18:11.38 | DelphiWorld | [TK]D-Fender: just je veux te dit qu'on doit oublié notre problem |
18:14.13 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
18:15.10 | [TK]D-Fender | DelphiWorld: Quelle probleme? |
18:15.20 | DelphiWorld | [TK]D-Fender: qu'on a créer |
18:15.52 | [TK]D-Fender | DelphiWorld: Le seul qu j'ai note a date etait que t'es resistent a l'idee d'apprendre |
18:17.25 | TobSnyder | [TK]D-Fender: any other ideas instead of relaxdtmf ? |
18:17.26 | DelphiWorld | [TK]D-Fender: less dit sa dans un PM |
18:17.48 | [TK]D-Fender | DelphiWorld: Je m'en souviens mem-pas no plus... |
18:17.51 | [TK]D-Fender | non* |
18:18.17 | [TK]D-Fender | DelphiWorld: Encore mieux de meme. |
18:23.14 | *** join/#asterisk UQlev (~Yuriy@212.50.99.8) |
18:24.19 | DelphiWorld | [TK]D-Fender: le problem c'est que je suis pas bien familiar avec le format d'asterisk |
18:26.42 | nort22 | Hello I'm new to this forum and have a question about asterisk |
18:26.52 | nort22 | has anyone seen WARNING file.c: Failed to write frame, after file.c: was Playing 'vm-intro.gsm' |
18:27.04 | nort22 | the dial plan steps appears to complete as normal and voicemail message is stored |
18:27.13 | nort22 | but i don't understand why i see "Failed to write frame" intermittently (1 every 400 calls) |
18:27.42 | [TK]D-Fender | DelphiWorld: Non, le problem c'est que vous-etait ici dupis plus qu'une an et demi et que tu refuse de LIRE. Il-y-a autant de guides partout incluant une livre pas mal officiel et meme t'es pas rendu au meme point qu'une personne avec un ti-peut effort peut reussir dans 2 jours |
18:27.50 | [TK]D-Fender | DelphiWorld: PAS D'EXCUSE |
18:28.14 | DelphiWorld | [TK]D-Fender: ok pas de problem |
18:28.32 | TobSnyder | http://pastebin.com/xs3q88vh |
18:29.32 | [TK]D-Fender | DelphiWorld: "je le connait pas" n'est pas un excuse pour ne pas avoir LU LE CRISE DE LIVRE EN 18 MOIS OU MOINS |
18:30.41 | TobSnyder | I'll leave now, if someone out there has some great hints, pls let me know |
18:31.10 | *** part/#asterisk zplinux (~zplinux@213.8.57.217) |
18:31.10 | *** part/#asterisk DelphiWorld (~Delphi@41.200.13.124) |
18:31.19 | TobSnyder | currently I am getting stuck with that problem |
18:46.20 | *** join/#asterisk mandragor (~ergudicsu@70.158.116.38) |
18:47.17 | mandragor | is there a way to setup a queue so that the call is not connected unless someone in the queue group answers the call? currently I can set it up so that when they enter the queue music plays in the background |
18:49.08 | [TK]D-Fender | mandragor: With excessive trickery & scripting, yes |
18:51.36 | mandragor | basically I have a service that sends calls to my company and I get charged if the call is answered so I have calls that are answered placed into the queue the customer then hangs up, I didn't deal with the customer but I still get charged |
18:53.14 | mandragor | [TK]D-Fender is there something you can point me to? |
18:53.42 | f00bar80 | I'm asking your opinion regarding a vps serving both asterisk/freepbx and websites hosting, the vps specs are as follows BW: 500 GB and Disk space 30 GB, any advice ?? |
18:54.15 | beardy | How many percent of your future profits are you offering here? |
18:55.59 | [TK]D-Fender | f00bar80: those mean nothing really |
18:57.34 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
18:59.20 | bmoraca_work | je n'en sais foutrement rien! |
19:00.31 | [TK]D-Fender | mandragor: There are a lot of pieces. Spawning new channels as place-holders in your queue, changing your agent dial's to local channels that will BridgeA() to the unanswered call's channel, etc |
19:01.02 | ManxPower | f00bar80, the standard advice is don't run Asterisk in a VM. If you insist on running Asterisk in a VM it might work, it might not work, it might be reliable, it might not be reliable, but don't expect support here. |
19:02.15 | citywok | especially if you don't own the hardware it's running on and have no control / no visibility in to server load. you could spend time debugging issues that come down to overloaded server/network |
19:03.05 | ManxPower | citywok, or the latency getting 1ms too high for clear audio |
19:03.13 | mandragor | [TK]D-Fender: Could I create a queue of max size 0? |
19:03.26 | [TK]D-Fender | ManfMeaning? |
19:03.33 | [TK]D-Fender | mandragor: What is "size"? |
19:03.39 | ManxPower | mandragor, why don't you just not answer the call? |
19:03.47 | citywok | i'm assuming you don't mean network latency, we run with 200-250ms of latency to one intl call center without issue, tho jitter sometimes plays hell. |
19:03.57 | [TK]D-Fender | ManxPower: He wants to queue incoming calls without answering until connected |
19:04.00 | ManxPower | many providers support early media |
19:04.09 | [TK]D-Fender | ManxPower: * isn't that smart |
19:04.16 | ManxPower | [TK]D-Fender, there isn't a noanswer option to queue? |
19:04.18 | citywok | Why? To avoid having to pay the provider for the hold time? |
19:04.26 | [TK]D-Fender | ManxPower: totla news to me if there is |
19:04.29 | citywok | Just force it to stay ringing inbound non stop? |
19:04.39 | [TK]D-Fender | ManxPower: I know DIAL has that.... |
19:04.39 | ManxPower | [TK]D-Fender, no wonder queues suck so much |
19:04.55 | bmoraca_work | mandragor: how much is this actually "costing" you? if it's pennies, I imagine the time being wasted here is costing more |
19:04.56 | ManxPower | [TK]D-Fender, so does background, read, etc |
19:05.16 | citywok | yea, unless you are talking thousands of minutes of hold time i wouldn't waste the energy |
19:05.21 | [TK]D-Fender | ManxPower: Really? I thought only playback did really... |
19:06.18 | mandragor | I was hoping it was something simple |
19:06.31 | mandragor | but you think it would require a very complicated script? |
19:06.39 | ManxPower | [TK]D-Fender, I just confirmed background does have a noanswer |
19:07.31 | [TK]D-Fender | ManxPower: I suppose that might work... I guess as soon as a digit is received that causes the telco to answer? Because* has to trigger it somehow... |
19:07.58 | ManxPower | mandragor, also you should know that MANY providers disconnect ringing calls after X seconds (120 is common) |
19:08.03 | [TK]D-Fender | ManxPower: Trying to think where Queue will ack that but not for prompts, etc, or if inbound DTMF automatically is processed by the telco |
19:08.22 | [TK]D-Fender | ManxPower: I guess he's also expecting a high service level |
19:08.40 | ManxPower | [TK]D-Fender, Heh, n00bs, always thinking their telco will be reliable. |
19:08.53 | breardo | ^^^^ learned about this TOO many times |
19:09.01 | breardo | NEVER EVER TRUST THEM!!!! |
19:09.44 | breardo | I especially love when Telco sales tells you they can do something.. and you sign the agreement.. and then the engineers say you cant do X |
19:09.51 | [TK]D-Fender | ManxPower: Actually... they SHOULD be the most reliable part of the equation... and if they suck.. you are kinda fucked on a macro-leevl anyway |
19:09.56 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
19:09.56 | ManxPower | We have a POTS line that Verizon has been trying to repair for TEN DAYS. |
19:10.16 | [TK]D-Fender | ManxPower: Should have gotten a T1 w/ SLA :p |
19:10.22 | mandragor | Manxpower: would I have to write a script to use the noanswer option? I am currently using fonality |
19:10.39 | [TK]D-Fender | ManxPower: Then you ARE fucked. You have NO control with that distro |
19:10.39 | bmoraca_work | well, there's your first problem |
19:10.42 | breardo | you're lucky..where I work, analog circuits have to be repaired by Qwest, but the ticket has to come from the state IT agency (MN OET).. its routine for it to take over two weeks for service to come on-site |
19:10.44 | ManxPower | mandragor, you mean the noanswer option that does not exist for queue (at least in the version of asterisk I'm using) |
19:11.00 | breardo | even though I could call qwest direct and they'd be here in 12 hours.. |
19:11.02 | breardo | nooooooo |
19:11.43 | ManxPower | [TK]D-Fender, I don't believe Verizon offers SLAs to CLECs |
19:12.07 | [TK]D-Fender | ManxPower: Life sucks, but rarely swallows ;) |
19:12.29 | breardo | i dont even care if life swallows..it could just smile once in awhile |
19:13.42 | ManxPower | [TK]D-Fender, what I think is *horrible* is that the PUC/PSC doesn't take complaints from CLECs and since the customer is a customer of the CLEC, not Verizon they can't file a complaint with the PUC/PSC either. |
19:14.32 | ManxPower | what is nice is that is a customer files a complaint with the PUC/PSC against the CLEC if the CLEC can show the issue is Verizon, the PUC basically ignores the complaint |
19:18.42 | bmoraca_work | verizon is just about the most aweful company ever |
19:19.09 | bmoraca_work | their prices are insanely high, too |
19:21.05 | *** join/#asterisk illizit (~illizit@c-76-110-117-217.hsd1.fl.comcast.net) |
19:21.48 | bmoraca_work | around here, if you're a customer, the worst company to deal with is Telekenex...if you're an outside tech (or competing telco trying to port-out), then Telepacific is the worst |
19:22.27 | jdwjr_ | I am seeing the following in our asterisk log WARNING file.c: Failed to write frame, after file.c: was Playing 'vm-intro.gsm' |
19:22.36 | jdwjr_ | Is this normal? |
19:23.11 | *** join/#asterisk TobSnyder (~Tobias@91-64-185-67-dynip.superkabel.de) |
19:24.52 | illizit | Hello, wondering if anyone knows what I am doing wrong. I am trying to setup Asterisk Realtime however, I am getting the following error: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available. |
19:25.01 | illizit | My configs are as follows: http://pastebin.com/RZYTfzH7 |
19:25.26 | illizit | asterisk 1.6.2.13 |
19:25.36 | illizit | astersik addons 1.6.2.2 |
19:26.40 | *** join/#asterisk rafael-ec (~rafael@200.25.197.106) |
19:29.20 | wcselby | o/ |
19:29.23 | *** join/#asterisk ghenry (~ghenry@pdpc/supporter/monthlybyte/ghenry) |
19:29.24 | ghenry | when a vendor talks about carrier mode and networking mode in a sip trunk, what do they mean? |
19:29.47 | ghenry | This is on a NEC IP card |
19:30.11 | bmoraca_work | ghenry: you'll have to ask him. that sounds like proprietary lingo to me. |
19:30.14 | ghenry | yep |
19:30.35 | bmoraca_work | ghenry: sight unseen, it seems like "carrier mode" would be trunk to a provider and "networking mode" might be a trunk to another PBX. |
19:30.44 | bmoraca_work | but that is complete conjecture |
19:30.51 | ghenry | They are asking if RFC2543 SIP is supported |
19:31.06 | bmoraca_work | only a NEC sales rep would know what the terms really mean |
19:31.10 | ghenry | yep |
19:31.23 | ghenry | 3261 is most current obviously plus additions |
19:31.38 | ghenry | and they are asking abotu 100rel which 1.4 doesn't have |
19:32.06 | ghenry | in fact it's disable on most softswitches I've used |
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19:44.06 | illizit | Hello, wondering if anyone knows what I am doing wrong. I am trying to setup Asterisk Realtime however, I am getting the following error: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available. |
19:44.34 | citywok | did you install the asterisk mysql module? |
19:44.48 | *** join/#asterisk oryxtec (~test@119.152.50.78) |
19:45.23 | oryxtec | hi all... can u use skype on asterisk ...with out using diguim solution.. coz that is really exp... |
19:45.31 | oryxtec | i have found this link http://nerdvittles.com/?p=680 |
19:45.32 | citywok | you asked that yesterday |
19:45.36 | citywok | nobody had an answer for you |
19:45.37 | oryxtec | has any one tried this? |
19:45.43 | illizit | yes, I did install the mysql module |
19:45.44 | drmessano | oryxtec, The other solutions are crap |
19:45.45 | oryxtec | why :( |
19:45.56 | drmessano | oryxtec, SFA is the only way to go |
19:46.09 | oryxtec | wht is SFA? |
19:46.14 | illizit | *CLI> realtime mysql status |
19:46.14 | illizit | general connected to asteriskconfig@localhost, port 3306 with username asterisk for 30 minutes. |
19:46.15 | drmessano | SKYPE FOR ASTERISK |
19:46.34 | drmessano | The one that is "really exp" |
19:46.49 | oryxtec | http://nerdvittles.com/?p=680 --------- soo this will not work? |
19:46.58 | leifmadsen | have you tried it? |
19:47.03 | leifmadsen | give it a shot and let us know |
19:47.32 | oryxtec | i have not tried it.. b4 i will try.. i need expert point of views on it |
19:47.40 | oryxtec | and you guys are experts |
19:47.45 | oryxtec | that is why i am asking you |
19:48.10 | drmessano | oryxtec, If you want to install X, then a 3rd party application, and route one concurrent call to the desktop client, there are ways |
19:48.17 | drmessano | Unreliable as hell ways |
19:48.49 | oryxtec | hummm |
19:50.21 | oryxtec | thanks :) |
19:50.56 | drmessano | $66 for a NATIVE channel driver that penetrates the Skype "bubble" is not bad at all |
19:51.09 | drmessano | Also comes with 1 G.729 license |
19:51.15 | oryxtec | humm |
19:51.57 | drmessano | and it supports Digium, who pays the fine devs that code the Asterisk you use for free |
19:52.14 | oryxtec | drmessano: i don't understand why skype sound quality is too good .. my servers are in UK as well.. |
19:52.30 | oryxtec | when i use my server on same link... sound is ok .. not bad |
19:52.42 | oryxtec | but when i use skype on same link.. sound is too good |
19:52.50 | oryxtec | why is that :S |
19:53.24 | drmessano | Skype is using a swell codec, for one |
19:53.52 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
19:53.54 | f00bar80 | ManxPower, i've mentioned it's a VPS hosting, virtual private server |
19:54.38 | oryxtec | wht is swell codec? |
19:55.40 | drmessano | oryxtec, awesome, cool, the bomb, sweet, swell, nice, the bees knees |
19:57.33 | oryxtec | :) |
19:58.48 | *** join/#asterisk erinspice (~erin@207.98.195.107) |
20:00.16 | illizit | city, any other ideas for my issue =D |
20:02.41 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
20:03.18 | ManxPower | f00bar80, unless you have proof otherwise, that is a VM |
20:05.10 | mandragor | how can I check if a given extension is busy? |
20:08.16 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.85) |
20:08.18 | *** join/#asterisk Chodorenko (~chodorenk@by.one.by) |
20:08.24 | Chodorenko | Hello All |
20:08.41 | Chodorenko | Please consult me by SRTP |
20:09.36 | Chodorenko | i read source code res_srtp and view "key exchange can be activated |
20:09.36 | Chodorenko | <PROTECTED> |
20:09.37 | Letoric | mandragor: Look into 'hint' |
20:10.28 | Chodorenko | i tried on asterisk 1.8.rc3 and no can secured dial from asterisk to client |
20:11.31 | Chodorenko | call from me go is secure, call from asterisk is no secure |
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20:19.25 | f00bar80 | ManxPower, so the only option is a "Dedictaed server" ?? |
20:20.31 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
20:24.54 | *** join/#asterisk Sheath (Smegma@unaffiliated/coil) |
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20:39.44 | *** join/#asterisk aut (~aut@office163.crdusa.net) |
20:41.00 | aut | i would like to set up a voip system for 10 users. it will be used for inbound customer service and regular outbound calls. can i just use any old sip trunk service? or do i need some kind of special connection to keep the latency low enough? |
20:41.21 | citywok | aut use whatever has good enough call quality for you. test it. |
20:41.23 | citywok | ~itsp |
20:41.23 | 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. |
20:41.39 | aut | citywok: can i use any isp so long as the latency/speed is enough? |
20:41.53 | aut | ~itsplist-us |
20:41.53 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
20:42.07 | citywok | you can use it... as long as it sounds good. |
20:43.06 | aut | so there are never really any guarantees, eh? i guess im asking whether you need qos guarantees to legitimately use voip for a small call center |
20:43.08 | citywok | If dial up sounds good enough for you, go for it |
20:48.47 | ManxPower | f00bar80, the only option if you want help from most of the people on this channel. You can do anything you please including using a turnip for your server. |
20:49.51 | ManxPower | You can get QoS if you get your internet service from your ITSP |
20:51.53 | p3nguin | That doesn't sound like something that happens very often. |
20:52.18 | aut | manxpower: right. isn't that the "proper" way to accomplish it? |
20:52.21 | citywok | p3nguin: depends on your carrier, getting XO to work w/you without a connection from them is impossible |
20:52.38 | citywok | aut: the proper way is whatever works with the right quality vs cost. |
20:52.39 | aut | manxpower: the problem is that the itsp that can provide local internet service isnt usually a well-known company |
20:53.19 | citywok | i get pretty high call quality running with bandwidth.com on my datacenters 100mbit link for instance, without a dedicated link just for voice from an itsp |
20:54.11 | citywok | i might get a slightly lower dropped call rate, or slightly better call quality but the number of complaints i get per 10,000 phone calls is so low it doesn't justify the cost / extra work required to bring in a pipe for just calls |
20:55.04 | aut | citywok: interesting |
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21:47.27 | docid | ok, slightly off topic and all, but i fuggure you gys might have experience with this, we've got many technologically limited users here at the office, we have a dieing meridian option-11, and a working stable asterisk install, using the standard meridian A II 4line phones, we want to move to asterisk before the meridian finishes dieing, so anybody have experence with what sip or iax phones would replace the meridian phones for useres obsti |
21:47.28 | docid | nante to learning anything new... |
21:47.38 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
21:54.49 | Letoric | docid: I use Polycom SoundPoint IP670 phones, they work well for us |
21:55.29 | docid | thats the one i was just looking at...but im thinking the price will make the boss say no, except mabey one for his desk |
21:55.39 | Letoric | heh |
21:55.47 | Letoric | well, you can go with the lower models of the same line |
21:55.49 | [TK]D-Fender | docid: Forget your old phones. They WILL be learning new stuff |
21:56.38 | docid | aye fender, im aware, but we have a few elders working here, so at least need 1 button pickup from transfer and hold |
21:57.00 | [TK]D-Fender | docid: Extremely unlikely |
21:57.02 | *** join/#asterisk poison (~poison@78-21-102-160.access.telenet.be) |
21:58.07 | poison | hi all, I'm createing some kind of escalation dial plan which rings number 1 on the list, and if that fails, it rings number 2, ... but the problem is that one of those numbers transfer directly to the voicemail (in case of a cell phone) then the call is answered and my application is broken |
21:58.13 | poison | is there a way to overcome this? |
21:58.33 | [TK]D-Fender | poison: "core show application dial" <- M() |
22:00.10 | poison | so there's something executed on the new called line, and there I can detect if it's a human voice or a voicemail? |
22:00.52 | docid | ohh, im curious on that one too |
22:04.23 | [TK]D-Fender | poison: Nothing resembling reliable. Promt them for CONFIRMATION. |
22:04.27 | [TK]D-Fender | prompt* |
22:05.17 | poison | so I should ask my users to press '1' if they are on some kind of voicemail? |
22:06.34 | [TK]D-Fender | poison: read the instructions again. You seem to have completely missed it |
22:07.23 | poison | I'm using asterisk 1.4.22-rc5 (from elastix), does your suggestion applies to that version too? |
22:10.06 | [TK]D-Fender | poison: To EVERY version. However unless you're already working up custom dialplan for this you're screwed. |
22:10.15 | Qwell | poison: Only 14 releases behind. |
22:12.19 | poison | <PROTECTED> |
22:12.31 | poison | sorry I don't quite get it yet |
22:14.57 | [TK]D-Fender | poison: Call the Cell, as the CELL SIDE to press "1" to ACCEPT the call and it will be CONNECTED |
22:17.56 | *** join/#asterisk csnook (~chris@138.210.3.1) |
22:20.32 | *** join/#asterisk b0gatyr_ (~b0gatyr_@adsl-2-204-53.mia.bellsouth.net) |
22:22.39 | poison | ok I'll check, thanks for the hints |
22:23.05 | poison | it's a pitty there's no special return code, I've read about machine detection modules but they don't see very reliable too |
22:34.37 | poison | is there a way to play back sound to the party I just called, and not to the caller (the one currently in my dialplan application) |
22:40.30 | [TK]D-Fender | [17:58]<[TK]D-Fender>poison: "core show application dial" <- M() |
22:58.00 | poison | hmz, [TK]D-Fender, I'm still struggling with your suggestion, I created a macro: macro-mustaccept, but it doesn't hangup the call if I don't press anything (http://pastebin.ca/1960488) |
22:58.35 | *** join/#asterisk [cannibalera] (~cannibale@201-35-198-209.fnsce703.dsl.brasiltelecom.net.br) |
23:00.07 | [TK]D-Fender | poison: Read the instructions again <- |
23:04.45 | [TK]D-Fender | Poincare: Hint : there are CHANNEL variables you should be paying attention to. |
23:23.04 | poison | is there a way to use the default outbound trunk in the Dial() program? |
23:24.52 | *** join/#asterisk Leddy (leddy@aqua.surgehost.net) |
23:26.48 | [TK]D-Fender | PoiThre is no such thing as "default". You tell it what to dial. The end. |
23:26.55 | [TK]D-Fender | poison: Exactly as it appears. |
23:29.23 | poison | well, I have [outbound-allroutes] where I configured all my outbound routes, and now in each Dial() command I have to specifiy which route I want to use instead of using the logic in [outbound-allroutes] |
23:34.38 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
23:36.00 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-103-152-61.cinci.res.rr.com) |
23:36.46 | *** join/#asterisk Hband_ (~Hband@ip68-2-140-46.ph.ph.cox.net) |
23:38.26 | [TK]D-Fender | poison: there is no default logic to a context name |
23:39.27 | [TK]D-Fender | poison: And Dial() calls the specific resource you tell it to. Again, this has what do do with a specific context? |
23:39.49 | *** join/#asterisk csnook (~chris@138.210.3.1) |
23:40.41 | poison | ok, it's just that whenever I update my outbound route information in freepbx, I have to update my script to make sure my primary voice provider is used to forward my Dial() calls to |
23:41.51 | *** part/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
23:43.58 | [TK]D-Fender | poison: FreePBX? it is NOT supported here and their junk is meaningless trash in here |
23:44.02 | [TK]D-Fender | ~freepbx |
23:44.02 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
23:44.04 | [TK]D-Fender | ^^^ |
23:44.55 | [TK]D-Fender | poison: If you're under it's yoke then things get messy, fast. Assuming that it doesn't force you to do everything manually if at all |
23:48.26 | poison | well, fortunately I'm creating my dialplan outside of freepbx (we just use it to enable non-technical ppl to do some basic stuff) |
23:54.19 | *** join/#asterisk aut- (~aut@oenf-nat-96-8-4-42.beyond.dssitech.com) |
23:55.28 | [TK]D-Fender | poison: Well there is no "routes". That means passing through their dialplan whose extra crap will FUBAR you |
23:56.01 | [TK]D-Fender | poison: So you are stuck doing it 100% manual for that segment. Do not expect to use any of their structures for this |