| 00:02.16 | norrec | idk if it does, i dont see it in my cdrs |
| 00:03.59 | Naikrovek | maybe that's just my wonky-as-all-getup trixbox setup |
| 00:04.37 | norrec | lol |
| 00:04.55 | norrec | well would u mind pbing the cdr conf files? |
| 00:05.09 | norrec | so i can see if urs is different than mine |
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| 00:33.08 | joobie | Naikrovek, sorry man - got caught up.. at work |
| 00:33.43 | joobie | Naikrovek, it's important because it conflicts with another firewall rule (got some fairly complex routing in that i dont want to complicate further to accomodate for this condition if possible) |
| 00:34.16 | Naikrovek | source ports aren't really a part of any firewall rules i've seen. destination ports are |
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| 00:34.40 | Naikrovek | the OS usually picks a port at random or via some selection mechanism I don't know about |
| 00:34.56 | Naikrovek | point being you can never count on a part going unused or consistently being used, for any service |
| 00:35.21 | Naikrovek | so, people don't base firewall rules on the source port. destination port, though, you can change |
| 00:35.44 | Naikrovek | s/part/port/ (two lines up) |
| 00:36.27 | Naikrovek | destination ports are static, though |
| 00:36.32 | Naikrovek | are you sure you're not thinking of source port |
| 00:36.34 | Naikrovek | ummmm |
| 00:36.39 | Naikrovek | DESTINATION port |
| 00:37.46 | Naikrovek | polycom may have it set up so that source port and destination port are the same. if you change the port asterisk listens on to 5061 or something, maybe the polycom will change its source port to that as well |
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| 00:40.38 | joobie | Naikrovek, they are apart of my firewall rules |
| 00:40.43 | joobie | we have some complex stuff going on.. |
| 00:40.48 | Naikrovek | i guess |
| 00:40.51 | joobie | I dont know if it's completely random |
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| 00:40.59 | joobie | the src port that is |
| 00:41.09 | joobie | some phones appear to be completely random |
| 00:41.12 | Naikrovek | i used to do firewalls for a living (well, cisco access lists for 300 customers) and never once did i filter based on source port |
| 00:41.14 | Naikrovek | but whatever |
| 00:41.15 | joobie | some seem to want to use 5060 as their src |
| 00:41.33 | Naikrovek | if you change the destination port number via ... you know what |
| 00:41.37 | Naikrovek | how do you configure the phones |
| 00:41.41 | joobie | src port filtering is common |
| 00:41.43 | joobie | such as FTP |
| 00:41.56 | joobie | but less common compared to dport filtering |
| 00:42.02 | Naikrovek | okay |
| 00:42.08 | Naikrovek | how do you configure the phones |
| 00:42.10 | joobie | i configure this via https:// |
| 00:42.15 | Naikrovek | web interface |
| 00:42.16 | joobie | but on a boot server |
| 00:42.24 | joobie | naa just use https as the transport for the boot server |
| 00:42.34 | joobie | used to use web interface but cant do what i need via that anymore |
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| 00:42.48 | Naikrovek | okay so you know all about sip.cfg and stuff |
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| 00:43.02 | Naikrovek | https:// ... do you generate the config files dynamically |
| 00:43.08 | Naikrovek | i think that could be useful |
| 00:43.19 | Naikrovek | have a webapp that generates the phone configs as they're requested |
| 00:43.31 | Naikrovek | anyway |
| 00:43.41 | Naikrovek | sip.cfg may have a setting for source port |
| 00:45.45 | Naikrovek | there's a fella at work i wanna prank. every 3rd or 4th time his phone boots i want his custom ringtone to be "i'm a barbie girl" or something |
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| 01:27.02 | BugKhaM | is the /etc/modprobe.d/zaptel required to start zaptel? |
| 01:27.25 | BugKhaM | thought it's done through an init script |
| 01:30.06 | joobie | Naikrovek, funny :) |
| 01:30.12 | joobie | Naikrovek, you should do it .. |
| 01:30.20 | joobie | Naikrovek, any idea what the sip.cfg setting is? |
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| 02:46.16 | norrec | does anyone know how to modify cdr output? |
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| 02:48.16 | p3nguin | /etc/asterisk/cdr_custom.conf |
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| 02:56.24 | rocksfrow | am i missing something I need to allow through my firewall for external sip connections? |
| 02:56.34 | rocksfrow | i have the appropriate ports unblocked, afaik.. |
| 02:56.47 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 02:56.54 | rocksfrow | my softphone is registering fine, and i can even dial out..pick up my cell phone, it shows i picked up..but no audio on either side |
| 02:56.57 | rocksfrow | and the call eventually drops |
| 02:57.01 | norrec | p3nguin: do you know where i can find some documentation on writing that file, cause i know nothing about cdrs... |
| 02:57.06 | rocksfrow | any clue? |
| 02:57.27 | rocksfrow | or where I could start debugging? everything looks normal in asterisk debug |
| 02:57.37 | norrec | where is ur asterisk server |
| 02:57.56 | rocksfrow | my office |
| 02:58.00 | rocksfrow | external |
| 02:58.03 | TJNII | ~sipnat |
| 02:58.04 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
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| 02:58.24 | norrec | is ur asterisk box natted? |
| 02:58.38 | rocksfrow | umm... |
| 02:58.45 | rocksfrow | yeah. |
| 02:59.04 | norrec | ok so its asterisk - firewall - pub internet - firewall - client |
| 02:59.05 | rocksfrow | i have the ports forwarded to the internal IP |
| 02:59.13 | rocksfrow | yerp |
| 02:59.30 | rocksfrow | and i have the appropriate ports setup, well..atleast I can make sip registration |
| 02:59.33 | rocksfrow | and make calls |
| 02:59.40 | rocksfrow | just no audio |
| 02:59.41 | norrec | ok, well you need to make sure asterisk know's its natted and that the client is natted |
| 03:00.24 | rocksfrow | okay, i guess i should give this a read through |
| 03:00.47 | rocksfrow | i don't want to mess up any of the internal phones though |
| 03:00.48 | BugKhaM | is the /etc/modprobe.d/zaptel required to start zaptel? or only an init script? |
| 03:00.48 | norrec | yeah look at #4 |
| 03:01.03 | rocksfrow | ah hah. |
| 03:01.12 | rocksfrow | norrec, let me confirm... this won't break inside connections through LAN |
| 03:01.21 | rocksfrow | probably a stupid question. |
| 03:01.30 | norrec | i cant promise anything, but in my experience no |
| 03:01.50 | TJNII | You can break internal phones if you really hose it up, but if you have two brain cells to rub together you should be fine. |
| 03:01.52 | rocksfrow | ugh..the howto link is a 404, lol |
| 03:02.05 | rocksfrow | i have port forwarding setup and working..but dono about STUN |
| 03:02.10 | rocksfrow | guess that's where the issue is |
| 03:02.37 | norrec | lol, hold on and i'll let u know which things u should prob set |
| 03:03.02 | rocksfrow | TJNII, lol, nice. |
| 03:04.12 | drmessano | Ouch.. If you're going to setup your tea maker and coffee maker on X10, make sure you remember which one is which |
| 03:04.41 | norrec | in global, set nat=yes, externip=external ip addr, and then nat=yes on each non-lan client |
| 03:04.44 | drmessano | HTCPCP 418: I'm a teapot |
| 03:04.48 | drmessano | :( |
| 03:05.50 | norrec | also set localnet = net.ip.addr/subnet.mask (ex 192.168.1.0/255.255.255.0) |
| 03:06.01 | TJNII | I was just about to call you on that. |
| 03:06.28 | drmessano | and throw a canreinvite=no in there |
| 03:07.12 | norrec | lol TJNII, i couldnt remember if netip/subnet was correct |
| 03:07.15 | norrec | had to look it up first lol |
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| 03:08.22 | norrec | does anyone have expeirence with modifying cdr outputs? |
| 03:08.22 | TJNII | norrec: No worries. I was quickly grepping my sip.conf before I said anything. |
| 03:08.30 | rocksfrow | eek, this is a freepbx box |
| 03:08.50 | TJNII | ~freepbx |
| 03:08.51 | 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 |
| 03:08.55 | rocksfrow | lol |
| 03:09.17 | TJNII | rocksfrow: Come back and see us if you ditch the GUIs, otherwise it's their way or the highway. |
| 03:09.43 | rocksfrow | TJNII, well, i'm actually on the verge of doing that... but not sure |
| 03:09.51 | TJNII | Do it. It's fun! |
| 03:09.59 | rocksfrow | i have a second box that i'm about to put asterisknow on for redundancy... and was considering going straight install |
| 03:10.36 | drmessano | Those settings for NAT awareness are the same |
| 03:10.52 | rocksfrow | drmessano, within an extension some of them don't exist |
| 03:10.53 | drmessano | Just need to place them in sip_general_custom.conf |
| 03:11.08 | rocksfrow | drmessano, i'm afraid if I manually edit my changes will just be overwritten |
| 03:11.11 | drmessano | Dont worry about the extension |
| 03:11.16 | drmessano | I just told you how to do it |
| 03:11.19 | rocksfrow | ohh... |
| 03:11.33 | rocksfrow | sip_general_custom..interseting |
| 03:11.50 | drmessano | nat=yes, externip=, localnet=, and canreinvite= |
| 03:11.50 | rocksfrow | it's empty :-p |
| 03:11.55 | drmessano | Yes it is |
| 03:12.04 | rocksfrow | well i don't see how these are static |
| 03:12.06 | drmessano | Just put those global settings in that file |
| 03:12.08 | rocksfrow | i mean global** |
| 03:12.11 | drmessano | They are |
| 03:12.17 | rocksfrow | i have internal clients as well |
| 03:12.28 | drmessano | Which is why you set localnet |
| 03:12.46 | rocksfrow | but, externip is not just on IP necessarily |
| 03:12.49 | drmessano | nat=yes is a behavioral setting based on the other parms |
| 03:12.53 | rocksfrow | this could be 20 different IPs |
| 03:12.54 | drmessano | nat=yes is a behavioral setting based on the other parms |
| 03:12.58 | rocksfrow | one SIP from each |
| 03:13.15 | rocksfrow | one** |
| 03:13.19 | drmessano | Your Asterisk box has 20 IPs? |
| 03:13.29 | rocksfrow | oh |
| 03:13.33 | rocksfrow | that hosts public ip |
| 03:13.35 | rocksfrow | sorry. |
| 03:14.28 | rocksfrow | so, localnet=10.0.0.0/255.255.255.0 look right? |
| 03:14.44 | drmessano | If that's your local subnet and it's mask |
| 03:15.11 | rocksfrow | well i just dont know if it should be 10.0.0.1/255.255.255.0 |
| 03:15.15 | rocksfrow | or is it supposed to be .0 |
| 03:15.21 | drmessano | Yes |
| 03:15.23 | drmessano | .0 |
| 03:15.40 | drmessano | But you're asking if that looks right.. |
| 03:16.04 | rocksfrow | after looking at my extensions in freepbx, they all currently have nat=yes |
| 03:16.05 | drmessano | Sure, or it could be wrong if that's not the subnet your boxes are on and you're not using a /24 |
| 03:16.08 | rocksfrow | i think that's a default value |
| 03:16.25 | drmessano | There is no default value |
| 03:16.46 | rocksfrow | well, i mean on the 'add extension' form within freepbx, the prefilled 'default' values, lol |
| 03:16.53 | rocksfrow | just saying, they're all set to nat=yes and have been working fine |
| 03:16.56 | rocksfrow | how come? |
| 03:17.17 | rocksfrow | maybe i can fix my fax problems by turning nat off on that extension? :-p |
| 03:17.27 | drmessano | No |
| 03:17.37 | maxagaz | hi |
| 03:17.50 | rocksfrow | drmessano, restart asterisk and give it a go? |
| 03:17.55 | drmessano | sure |
| 03:19.21 | maxagaz | I still have the problem of a zero added between the extension set to call outside and the number I need to call, I can't see this zero in the logs not in the console, does someone have an idea ? |
| 03:19.48 | *** join/#asterisk spenguin[work] (~penguin@59.162.86.164) |
| 03:19.52 | spenguin[work] | TEST |
| 03:19.54 | p3nguin | ~freepbx |
| 03:19.55 | 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 |
| 03:20.22 | rocksfrow | duuuuude |
| 03:20.33 | rocksfrow | drmessano, norrec, etc...you da mannn |
| 03:20.36 | rocksfrow | damn. |
| 03:20.39 | rocksfrow | that easy lol |
| 03:21.12 | rocksfrow | so i had the port forwarding working fine, but the externip i think is what i was missing |
| 03:21.18 | drmessano | Yes, I am well aware. I get lots of high 5's from the bro's and "oh my god, no man has ever.. " from the ladies. |
| 03:21.36 | rocksfrow | kinda scary i just have to wonder if everything is working in the office still |
| 03:21.47 | rocksfrow | drmessano, so i should probably update the internal extensions to nat=no? or does it matter |
| 03:21.52 | drmessano | NO |
| 03:22.07 | rocksfrow | whoa |
| 03:22.14 | drmessano | I told you twice |
| 03:22.25 | TJNII | drmessano: International man of mystery, intrigue, and telephone repair. |
| 03:22.36 | drmessano | NAT=yes is an "ALLOW" based on other conditions, one of those being localnet |
| 03:23.10 | rocksfrow | okay. |
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| 03:25.02 | norrec | rocksfrow: + ive set lan clients as nat w.o using localnet and not had any issues |
| 03:25.22 | rocksfrow | norrec, yeah thx |
| 03:25.34 | rocksfrow | i think the externip was what i needed |
| 03:25.38 | drmessano | You would only change that to "never" if you needed some device to ALWAYS use the public IP |
| 03:25.52 | rocksfrow | gotcha. |
| 03:26.01 | drmessano | as in, a LAN device |
| 03:26.07 | drmessano | on the 10.0.0.0 subnet |
| 03:26.09 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 03:26.37 | drmessano | as in "I know you're local, but screw you, use the external IP for call setup because I hate you, bro" |
| 03:27.01 | drmessano | Im sure there's a reason under some condition to do that, but I have never not set extensions to nat=yes |
| 03:27.38 | rocksfrow | drmessano, i guess that explains the default setting in freepbx |
| 03:27.56 | rocksfrow | thanks very much for the explanation |
| 03:28.03 | rocksfrow | sorry to have you say it three times. :-/ |
| 03:28.05 | drmessano | Correct, because it acts more like an allow and not forcing some condition, it's a safe default |
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| 03:33.01 | Naikrovek | joobie: no idea what the setting is, but look for 5060 in sip.cfg and see if one look like it might be it |
| 03:33.24 | joobie | tried that :/ |
| 03:33.55 | joobie | .. unless im plain outright missing it |
| 03:34.44 | joobie | voIpProt.local.port |
| 03:34.50 | joobie | unless it's that? but not clear by the docs |
| 03:34.58 | Naikrovek | that sounds like it |
| 03:35.10 | joobie | Local port for sending and receiving SIP signaling packets. |
| 03:35.11 | joobie | If set to 0 or Null, 5060 is used for the local port but it is not advertised in the SIP signaling. |
| 03:35.11 | joobie | If set to some other value, that value is used for the local port and it is advertised in the SIP signaling. |
| 03:35.21 | joobie | ergh - scuse the long paste.. thought it would be a one liner. |
| 03:35.26 | joobie | that above is the description for that option tho |
| 03:35.35 | Naikrovek | won't worry about it. change it - i bet it changes the source port |
| 03:36.00 | Naikrovek | or maybe it changes where it listens for SIP connections from asterisk |
| 03:36.03 | Naikrovek | won't know til you try |
| 03:36.11 | Naikrovek | ah yes you will |
| 03:36.16 | Naikrovek | it says "Sending and receiving" |
| 03:36.20 | Naikrovek | there's your answer |
| 03:36.30 | Naikrovek | that is how you change the source port |
| 03:38.42 | drmessano | Why would set up firewall rules based on SOURCE port? |
| 03:38.49 | Naikrovek | that was my question |
| 03:38.53 | Naikrovek | he says its common |
| 03:39.02 | Naikrovek | i disagree but i'm no firewall expert so i moved on |
| 03:39.43 | joobie | i lost you Naikrovek - is voIpProt.local.port the answer? |
| 03:39.45 | *** join/#asterisk Kumbang (~kumbag@rusnas.paume.itb.ac.id) |
| 03:39.54 | drmessano | I'm sure web browsers are pissing them off then |
| 03:39.58 | joobie | it's very common |
| 03:40.03 | joobie | just look at the FTP protocol |
| 03:40.17 | joobie | it's just not as common as dport |
| 03:40.37 | joobie | the reason im doing it is i have a connection at this site that i can make free local / national calls from |
| 03:41.07 | joobie | so ive some funky nat setup so that i can route an asterisk box at a another site via VPN+NAT and emulate the connection as if it was originated locally |
| 03:41.47 | joobie | could just substitute this with another asterisk box at this site.. |
| 03:42.34 | joobie | my old job, we had *the* most complex iptables setup going across 3 different datacenters |
| 03:43.30 | joobie | it sucked when you wanted to amend things in the ruleset - easy to break something else - but at the same time steps up your knowledge of firewalling / nat / routing 10 fold. |
| 03:44.53 | joobie | BTW Naikrovek, that voIpProt.local.port set the src port correctly :) Thnx |
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| 06:05.39 | ChannelZ | Can GoDaddy suck any more ass? |
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| 06:17.58 | p3nguin | not easily |
| 06:18.09 | p3nguin | They've done a pretty bang-up job already. |
| 06:19.27 | p3nguin | I bought my first domain name with them in 2003, and they weren't that bad at that time. |
| 06:19.40 | p3nguin | No, actually, I transferred in to them. |
| 06:20.13 | p3nguin | from some junky-ass registrar that didn't even let me control my DNS. |
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| 06:26.49 | carrar | If they don't have the ability to do glue records they aren't worth using |
| 06:28.06 | carrar | IPv6 glue records that is |
| 06:29.09 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 06:33.53 | Milad | is there anyway to add sip header in 1.4 ? it seems remove after 1.2, right ? |
| 06:34.43 | carrar | SipAddHeader() |
| 06:35.26 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 06:35.34 | Milad | # Asterisk func SIPAddHeader: Typically used to set Alert-Info information, e.g. ring tone .wav files. (1.2) |
| 06:35.55 | Milad | ow it mean add from 1.2 ! |
| 06:35.56 | Milad | tnx |
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| 06:40.37 | aceio | how do i turn on callerid |
| 06:41.59 | aceio | i am waching cli console Accepting call from '' to 'xxxxx' on channel 0/1, span 1 |
| 06:42.41 | aceio | and i would like too see the incoming numbers |
| 06:47.03 | ChannelZ | you have to print it... NoOp(${CALLERID(all)}) |
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| 06:57.14 | aceio | thx ChannelZ |
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| 06:59.19 | maxagaz | I still have the problem of a zero added between the extension set to call outside and the number I need to call, I can't see this zero in the logs nor in the console, does someone have an idea ? |
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| 07:00.19 | carrar | Sounds like a dialplan typo |
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| 07:00.52 | carrar | how do you know it's there if it's not in the logs |
| 07:00.58 | ChannelZ | yeah need to see your Dial() and a call |
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| 07:46.32 | maxagaz | carrar, I'm in Beijing, the server is in Shanghai, I compose 9 followed by the number I need to call, for mobile phone outside shanghai, I have to compose 0 first |
| 07:47.57 | maxagaz | carrar, so, I did the following tests: I called my mobile phone number in Beijing by just composing 91580135**** and it works, while it should be 9015801**** |
| 07:49.04 | maxagaz | carrar, when I compose shanghai's mobile, it says no need to compose 0 before, and then it doesn't work |
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| 08:05.13 | carrar | Sounds like a dialplan issue |
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| 08:45.13 | tengulre | hi,all |
| 08:45.41 | tengulre | anybody build call center under asterisk ? |
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| 09:01.13 | maxagaz | carrar, I can't see any error in my dialplan... |
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| 09:04.40 | Polysics | hello |
| 09:04.55 | Polysics | asterisk 1.6.1 - can i implement some sort of "single user queue"? |
| 09:05.13 | Polysics | aka. if i direct call an user and he is busy, can i put the call in a sort of "waiting list"? |
| 09:05.58 | Polysics | a problem might be that users are also part of queues |
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| 09:28.51 | kaldemar | Polysics: make a queue with a single member |
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| 09:31.53 | Polysics | kaldemar, then i would have each user in two queues |
| 09:32.02 | Polysics | does that work? |
| 09:32.11 | kaldemar | there's no problem with that. |
| 09:32.37 | Polysics | one other thing: i have two users, which are also members of queues, that are talking to each other |
| 09:32.58 | Polysics | user A is member of queue 1. let's say i have a third user, member of queue 1 too |
| 09:33.27 | Polysics | when someone calls queue 1, the queue does not ring user C, instead it waits for A to finish the call, then calls user A |
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| 09:38.51 | Polysics | any idea why that happens? |
| 09:40.39 | Polysics | btw, i think i could just turn on call waiting on my extensions, if i could find the syntax anywhere |
| 09:40.42 | kaldemar | because of your queue strategy or some other configuration? |
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| 09:42.43 | Polysics | strategy is rrmemory - but shouldn't the queue call the first available member' |
| 09:42.45 | Polysics | ? |
| 09:43.10 | Polysics | i mean, the member the queue "wants" to call is busy, but there is one that is not |
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| 09:58.05 | Polysics | i think the only solution could be pausing the user from queues |
| 09:58.54 | Polysics | then on the hangup event i unpause him |
| 09:59.08 | Polysics | i suppose the queue does not "know" the user is being called |
| 09:59.41 | Gopal | Polysics: try to change the queue ringing strategy as round robin |
| 10:00.34 | *** join/#asterisk jkroon (~jkroon@dsl-244-37-97.telkomadsl.co.za) |
| 10:01.16 | jkroon | hi guys, got a problem with MOH not acting as it should. |
| 10:01.48 | jkroon | transfer happens, when user presses "transfer" the caller goes to MOH, but as soon as it's then supposed to start ringing (destination channel signals ringing) it's just silent. |
| 10:02.15 | jkroon | asterisk 1.6.1.18, confirmed with incomin channels of IAX/2 and DAHDI, transferring channel is SIP, destination tech doesn't matter. |
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| 10:03.24 | Gopal | jkroon: codec might be problem... |
| 10:03.38 | jkroon | Gopal, explain ? |
| 10:04.11 | Gopal | jkroon: if you have different codec in the destination when compared to transferring line |
| 10:04.19 | Gopal | jkroon: that could be the prob |
| 10:04.31 | Gopal | jkroon: are you transferring to same network or to outside network |
| 10:05.52 | jkroon | well, in this particular case it's incoming on DAHDI (PRI/T1) -> SIP. SIP then puts DAHDI on hold by pressing the transfer button (and MOH plays), at this point she dials the new destination (usually another SIP extension on the same network, sometimes an external DAHDI destination), and as soon as that phone starts ringing the caller gets silence. |
| 10:06.33 | Polysics | Gopal, no, the same problem persists |
| 10:06.47 | Polysics | i would say it is a matter of pausing users |
| 10:09.32 | Gopal | jkroon: have you given transfer = yes in chan_dadhi.conf? |
| 10:10.20 | Polysics | when a user is unpaused from queues, and he is hte only agent in a queue, does the queue get registered as "empty"? |
| 10:10.30 | jkroon | Gopal, was set to no. what exactly does it change? |
| 10:10.36 | jkroon | Polysics, no. |
| 10:10.45 | jkroon | this is considered a feature, not a bug :( |
| 10:11.23 | Gopal | Polysics: yes |
| 10:11.37 | Polysics | jkroon, which means i could pause users taht are engaged in a "direct" call, and calls to the queue would properly "wait"? |
| 10:11.42 | Polysics | Gopal, oh lol :-) |
| 10:12.20 | Gopal | Polysics: not for your question just |
| 10:12.35 | jkroon | Polysics, when a user is unpaused there will be >0 agents in the queue. there is a queue option that will prevent app_queue from sending calls to in-use consumers from the queue. |
| 10:12.59 | jkroon | ringinuse=no <-- should do what you want. |
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| 10:14.21 | Polysics | jkroon, ringinuse is already set (to 0, not to "no" since queues are in realtime) |
| 10:14.54 | jkroon | ok well, works for me so far. |
| 10:14.55 | jkroon | Gopal, what does transfer = yes change? |
| 10:15.41 | Gopal | jkroon: it will transfer for flash-hook http://www.voip-info.org/wiki/view/chan_dahdi.conf |
| 10:15.57 | jkroon | Gopal, you don't understand the problem then. |
| 10:16.07 | jkroon | that transfer i want on no |
| 10:16.24 | jkroon | the issue is that the incoming caller (from dahdi) doesn't hear ringing when the sip side transfers. |
| 10:17.07 | jkroon | you don't want people that call in to your system to be able to initiate transfers ... imho. |
| 10:17.08 | Gopal | jkroon: ok |
| 10:17.27 | Gopal | jkroon: ok I understood |
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| 10:18.06 | Gopal | jkroon: can you able to transfer a call from SIP to SIP without using dahdi |
| 10:18.06 | jkroon | o.O |
| 10:18.24 | ampilogov_a_ | hi there :) |
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| 10:20.36 | Polysics | jkroon, basically, my queues do not honor ringinuse |
| 10:20.38 | jkroon | ok, let me rephrase again. some external to my asterisk system user calls in coming in off my DAHDI channel coming from external. This goes into a Queue() that gets answered by a SIP/ channel. So now I have a bridge between DAHDI/x and SIP/y. At this point y presses "transfer" (putting x on hold) and dials a new number (let's say z), which ends up going to SIP/z, now as soon as SIP/z signals ringing DAHDI/x should hear ringing, but in |
| 10:20.39 | jkroon | stead it gets silence. |
| 10:20.43 | ampilogov_a_ | can you help with asterisk? I have trixbox system with SIP phones. There is FOP in the trixbox where you can connect 2 SIP phones by drag&drop. I want to do the same but programmaticaly on PHP (for examp php-agi) |
| 10:21.17 | Polysics | ampilogov_a_, you need to use AMI originate |
| 10:21.35 | Polysics | you can do that from php-agi |
| 10:21.43 | ampilogov_a_ | does it work for ineranl SIP numbers? i can't dial external numbers |
| 10:21.56 | Polysics | just be careful as Originate is a bit difficult at first |
| 10:22.02 | Polysics | it sure does |
| 10:22.27 | ampilogov_a_ | ooh, allright, will read documentation |
| 10:23.20 | Gopal | jkroon: but in the otherend SIP/z there is a ringing, only in dhadi/x there is no ring back tone, correct? |
| 10:23.30 | jkroon | correct! |
| 10:23.51 | Gopal | jkroon: what is your tx and rx gain value ? |
| 10:25.38 | Gopal | jkroon: hey whats your dialroute you have given to reach the SIP/z extension? |
| 10:25.56 | Gopal | jkroon: you need to add "r" in the dial route to hear the ringback tone |
| 10:26.26 | jkroon | Dial(SIP/z) |
| 10:26.49 | Gopal | jkroon: try this Dial(SIP/z,Tr) |
| 10:26.55 | jkroon | why would I need an explicit r? surely the fact that I receive Ringing() from SIP/z will cause asterisk to pass that back to the DAHDI channel? |
| 10:27.29 | Gopal | jkroon: to hear the ringback tone you need to give r for TDM/PSTN |
| 10:27.38 | jkroon | Gopal, surely you mean Dial(SIP/z,,Tr) ? |
| 10:27.45 | Gopal | jkroon: yes |
| 10:27.59 | Gopal | jkroon: I faced the same prob in PSTN after giving r it worked for me |
| 10:28.16 | jkroon | <PROTECTED> |
| 10:28.17 | jkroon | <PROTECTED> |
| 10:28.24 | jkroon | just the r then. |
| 10:29.12 | ampilogov_a_ | hmm, i'm not shure i've done all correct, so is there simple test phone using php-agi script, for instance, to call my SIP phone? |
| 10:29.13 | ampilogov_a_ | require_once('/var/lib/asterisk/agi-bin/phpagi-asmanager.php'); $number = '222'; $asm = new AGI_AsteriskManager(); if($asm->connect('localhost:5038', 'admin', 'amp111')) { echo "connect"; $call = $asm->send_request('Originate', array('Channel'=>"SIP/$number", 'Context'=>'default', 'Priority'=>1, 'Callerid'=>$number)); |
| 10:33.42 | jkroon | Gopal, it works, however, i've got a concern regarding it, what happens if the phone isn't available? does it wait until receiving ringing from the new SIP/ channel before sending ringing or does it just send it immediately after receiving 100 Trying or even before that? |
| 10:34.53 | Gopal | jkroon: if the phone is not avaialable it will return back some different tone |
| 10:35.28 | jkroon | i'm still sceptical. i recall having had false rings with that option before. |
| 10:36.35 | Gopal | jkroon: by default it has to ring without "r" option, I too tried a lot and did the same... but need to check is it giving false ring...:D |
| 10:37.19 | Gopal | jkroon: if SIP phone is not there defnitely it will not ring as I suspect... |
| 10:37.32 | Gopal | jkroon: since the SIP registration will not be there... |
| 10:48.18 | jkroon | Gopal, then for internal extensions it should be OK. |
| 10:48.58 | Gopal | jkroon: ok |
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| 10:49.34 | jkroon | however, my case that I recall being problematic was a system where I'd receive a call via IAX/2 or SIP, and then call out to an upstream via SIP doing Dial(SIP/upstream/${EXTEN},,r) ... this would start ringing immediately, and then 3 seconds later drop the call when hitting a 404 or something. so I dropped the r and the problem went away. |
| 10:49.45 | oktay | hello. anybody have the thomson 780wl ? (or know whether the antenna is removable?) |
| 10:52.24 | Gopal | jkroon: so now you are not using r? |
| 10:52.34 | jkroon | not on that particular switch no. |
| 10:52.49 | jkroon | that r and m option has been confusing the crap out of me. |
| 10:53.05 | jkroon | but that switch also doesn't deal with transfers, ever. |
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| 11:01.02 | oktay | it is not removable. thanks for listening :) |
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| 11:17.34 | Gopal | jkroon:ok |
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| 11:20.32 | Bartockbatz | hey folks - a fast and easy way to have the asterisk server email the last 5 incoming call records - so far, I am using a shell script to awk out the last 5 records - |
| 11:21.09 | Bartockbatz | instead of reinventing the wheel, I wanted to check if there are any tools available for 1.4xx |
| 11:21.35 | jkroon | tail instead of awk? |
| 11:21.49 | jkroon | unless you're re-formatting .... :p |
| 11:22.03 | Bartockbatz | oh yeah - used tail |
| 11:22.10 | Bartockbatz | let me show you what I ue |
| 11:22.12 | Bartockbatz | use |
| 11:22.39 | Bartockbatz | tailf -n 5 /asterisk/log/csv_file |
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| 11:23.10 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:23.18 | Bartockbatz | awk -F',' ( I would have to look at the fields that I grabbed - basically the incoming call, the time and date) |
| 11:23.32 | Bartockbatz | then redirect that to a date stamped file |
| 11:23.41 | Bartockbatz | and email the text to an email address |
| 11:24.22 | Bartockbatz | it works, but I was wondering if there is a 'way to build a better mousetrap' |
| 11:24.37 | jkroon | depends on what you want to do but it sounds about right. |
| 11:25.27 | Bartockbatz | so, no fancy tools that I can grab that will save me the scripting for formatting the output? not that I mind doing it, but I like to work smarter, not harder |
| 11:26.29 | Bartockbatz | I saw a tool called 'Phone Genie' - but I am wondering if it would suit my needs |
| 11:27.57 | Bartockbatz | Well, I am going to format it to make it look pretty, just to keep the client happy ad then look around for a tool |
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| 11:45.34 | Gopal | what are all the ports to open if i have a firewall? |
| 11:45.34 | Gopal | TCP and UDP? |
| 11:46.50 | beefpastry | Gopal: http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules |
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| 11:51.19 | hajkym | hi can me sombody help with this? |
| 11:51.24 | hajkym | http://pastebin.org/257506 |
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| 11:51.58 | hajkym | i want create one user which can only make call and user which can only receive call |
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| 12:03.12 | Gopal | beefpastry: do you have any idea about RTCP.. do we also enable port for that... |
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| 12:24.00 | thomas | Aloha! |
| 12:24.33 | ThoMe | Is it posible with the asterisk-console or API/manager? check if the user logged in? |
| 12:25.40 | [TK]D-Fender | ThoMe: What "user"? Logged into what? |
| 12:26.15 | ThoMe | [TK]D-Fender: login via SIP in asterisk, sorry. |
| 12:26.53 | tzafrir | ThoMe, sip show peers? |
| 12:26.57 | tzafrir | sip show registry? |
| 12:27.21 | [TK]D-Fender | tzafrir: Too obvious.... |
| 12:27.35 | ThoMe | [TK]D-Fender: what is better? ;) :P |
| 12:27.42 | ThoMe | tzafrir: i think show peers or? |
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| 13:39.13 | rrb3942 | Are their any characters that cannot be used in a SIP secret? or should be avoided? |
| 13:40.05 | ManxPower | rrb3942, Is there a specific character that you have in mind? |
| 13:41.13 | rrb3942 | Mostly wondering if any special characters will cause a problem |
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| 13:42.23 | rrb3942 | I am creating a script to generate a csv for use with the freepbx bulk extension module and right now it just makes a passwords from any printable characters |
| 13:44.18 | rrb3942 | So ',' is a problem since it is a csv, but I don't know if maybe '\/#+' would be an issue for the sip secret in asterisk |
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| 13:50.51 | krion | anyone using thomson st2022 with fw v4.69 ? |
| 13:51.13 | krion | i know it's not asterisk related but i have quality problem with it... |
| 13:51.21 | krion | wonder if anyone already fix it |
| 13:56.14 | ManxPower | I suspect # is the only printable char that is not supported. " is not a problem in CSV |
| 13:59.04 | rrb3942 | Alright, sounds good |
| 13:59.46 | rrb3942 | Ill add in quotes for good measure as well |
| 13:59.48 | rrb3942 | thanks |
| 14:00.09 | ManxPower | you will need to escape the " of course |
| 14:00.52 | rrb3942 | yep |
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| 15:20.50 | LemensTS | does nat=yes send keep-alives or options? |
| 15:21.02 | pabelanger | LemensTS: no |
| 15:21.44 | pabelanger | LemensTS: only real solution is to lower your SIP registration / qualify timers for nat-keepalive |
| 15:22.15 | pabelanger | or port-forwarding on your router |
| 15:22.35 | LemensTS | this is for phone device |
| 15:23.08 | pabelanger | LemensTS: Same rules apply |
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| 15:23.30 | LemensTS | That is done with qualify=XXX right |
| 15:23.42 | pabelanger | LemensTS: correct |
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| 15:24.21 | LemensTS | ok. im having a problem with it going unreachable. I had it on qualify=yes nat=yes....what ms you think i should try? |
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| 15:27.59 | pabelanger | LemensTS: 200ms is the default |
| 15:28.11 | pabelanger | LemensTS: Also, what version of Asterisk you using? |
| 15:28.26 | LemensTS | 1.4.25.1 |
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| 15:29.58 | pabelanger | LemensTS: you should be fine in the 1.4 branch |
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| 15:34.45 | LemensTS | pabelanger: http://pastebin.com/1hMDSFdr does retrasmitting like this mean it is sending options to the device, but the device isn't responding so it is trying again (retransmitting) |
| 15:35.21 | LemensTS | it trys it 4 times than it destroys the sip deialog |
| 15:35.40 | pabelanger | ~sipnat |
| 15:35.41 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:35.48 | pabelanger | LemensTS: ^^^ |
| 15:36.05 | pabelanger | LemensTS: Likely a configuration issue |
| 15:36.30 | Belgarath | use dafaultexpirey |
| 15:36.35 | Belgarath | and lower the limit to 10 sec |
| 15:36.47 | Belgarath | then go up until you find highest working value |
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| 15:38.27 | rrb3942 | some routers also have udp session timeout options, increasing those to several minutes can help sometimes |
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| 15:46.10 | Znuff | NGEN |
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| 15:54.37 | puzzled | tzafrir: about http://docs.tzafrir.org.il/dahdi-linux/README.html#_oslec : even on CentOS 5.5 with kernel 2.6.18 do you still need to take a newer kernel like 2.6.28 as it says in your doc? |
| 15:54.40 | *** join/#asterisk vk4akp (~Ken@c114-77-251-186.ipswc3.qld.optusnet.com.au) |
| 15:55.17 | vk4akp | HI. Looking for some help with the ChanSPy command. |
| 15:55.24 | tzafrir | puzzled, "need newer kernel" is for getting oslec from that kernel |
| 15:55.36 | tzafrir | If you don't, just use any kernel (supported by DAHDI) |
| 15:55.48 | puzzled | tzafrir: got it. thanks |
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| 16:00.31 | vk4akp | exten => _88XXXX,1,Chanspy(SIP/${EXTEN:2}|b) |
| 16:01.00 | vk4akp | How can I change this so teh number entered after the 88 can be of variable length. Like 1-4 long instead of having to be 4 long? |
| 16:01.11 | LemensTS | if there is a defaultexpirey, why is there a maxexpirey? |
| 16:01.32 | LemensTS | i dont see how it could be a range |
| 16:03.13 | *** join/#asterisk shader (~40846872@gateway/web/freenode/x-tlrpyxnujngkhutf) |
| 16:04.51 | shader | do I need to implement things like congestion tones myself when I dial out to an external number, or will the remote end handle that? |
| 16:06.31 | idespinner | vk4akp, i would just make 4 entries: _88x, _88xx, _88xxx, _88xxxx |
| 16:07.22 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:08.01 | vk4akp | I don't seem to be able to make this feature work anyhow. |
| 16:08.03 | [TK]D-Fender | exten => _88., <- variable length |
| 16:08.17 | vk4akp | Have you used this ChanSpy feature? |
| 16:08.28 | idespinner | [TK]D-Fender, except that includes 1-infinity digits |
| 16:08.29 | [TK]D-Fender | vk4akp: What ver of * are you using? |
| 16:08.36 | idespinner | i think he wants 1-4 digits only |
| 16:08.42 | vk4akp | Yes [TK]D-Fender, I ended up doing it with the DOT . thanks. |
| 16:08.49 | [TK]D-Fender | idespinner: My statement remains entirely accurate :) |
| 16:08.57 | vk4akp | Um. I am using the special one for APP_RPT |
| 16:09.09 | [TK]D-Fender | vk4akp: Now try expressing that with NUMBERS. |
| 16:09.20 | vk4akp | Asterisk SVN--r588M built by root @ shazam on a i686 running Linux on 2010-04-01 14:19:13 UTC |
| 16:09.40 | [TK]D-Fender | vk4akp: What branch is that from? |
| 16:09.53 | vk4akp | YOu would have to tell me how to find out. |
| 16:10.09 | vk4akp | It's a special one built for APP_RPT it runs into a aradio network. |
| 16:11.26 | *** join/#asterisk chazzam (~chazz@173-24-238-25.client.mchsi.com) |
| 16:11.39 | vk4akp | I think it is 1.4.somethign with stuff added to fix bugs. |
| 16:12.43 | shader | vk4akp: were you the one that compiled it? |
| 16:12.50 | vk4akp | No. |
| 16:13.04 | *** part/#asterisk LemensTS (~LemensTS@adsl-70-238-143-123.dsl.stlsmo.sbcglobal.net) |
| 16:13.05 | vk4akp | the APP_RPT / All-STar guys do all that. |
| 16:13.09 | vk4akp | I am just a user. |
| 16:13.16 | shader | ok |
| 16:13.28 | vk4akp | Well. I guess technically it was compiled on our system. |
| 16:13.40 | shader | but you didn't do the installation yourself |
| 16:13.43 | vk4akp | Its a special version for Amateur Radio use. |
| 16:13.56 | vk4akp | Originally I installed it with a alot of problems. |
| 16:14.08 | vk4akp | But this release was installed by one of the nice ALL_Star guys |
| 16:14.23 | vk4akp | They updated the system for me a month or so back. |
| 16:14.40 | shader | [TK]D-Fender: if you dial out to an external number, say via a sip service provider, do they provide features like busy tones, or do I have to implement that myself? |
| 16:14.56 | shader | vk4akp: ok |
| 16:15.15 | shader | vk4akp: did they check out the source onto your system? it might be in /usr/src |
| 16:15.18 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 16:15.23 | shader | in which case you could probably figure out the version |
| 16:15.27 | vk4akp | So anyhow. I am looking for ways to monitor different extensions on teh PBX. |
| 16:15.39 | vk4akp | I currently have issues where the system is being abused and I need a way to monitor. |
| 16:17.00 | *** join/#asterisk Ta^3 (~tacvbo@189.146.181.250) |
| 16:17.08 | shader | what kind of monitoring are you looking for? |
| 16:17.21 | shader | ChanSpy and ExtenSpy seem to be for recording calls |
| 16:19.38 | p3nguin | ExtenSpy() spies based on the extension, ChanSpy() spies on the channel. Neither are for recording. |
| 16:20.33 | p3nguin | MixMonitor() and Monitor() are used to record calls. |
| 16:22.01 | shader | fine, but they seem to be for listening in on a channel, with the option for recording |
| 16:22.05 | *** join/#asterisk retentiveboy (~pdugas@atl.pra-corp.com) |
| 16:22.19 | shader | which may or may not be what vk4akp is looking for |
| 16:22.29 | ManxPower | vk4akp, "core show applications" is your friend. |
| 16:22.52 | ManxPower | (might even be a friend with benefits....of knowing Asterisk if you play your cards right |
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| 16:23.50 | p3nguin | Yeah, I didn't mean to mislead by saying the aren't for recording. I should have said their main purpose is not recording. |
| 16:24.25 | ManxPower | shader, with VoIP (any protocol) or ISDN (any form) you get back a status code, it is up to you to play Busy, Congestion or Whatever. |
| 16:25.29 | shader | ok |
| 16:25.45 | shader | how do you have yours set up? |
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| 16:35.39 | BANSAL | can anybody suggest me any ebook or web link which can provide me complete documentation about asterisk ? |
| 16:35.51 | Qwell | ~book |
| 16:35.52 | 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 |
| 16:36.39 | vk4akp | Sorry back. |
| 16:36.46 | vk4akp | ExtenSpy might do what I need then. |
| 16:39.56 | shader | vk4akp: what do you need? |
| 16:41.00 | vk4akp | The ability to monitor an extension that someone is on. |
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| 16:58.34 | Kevin` | vk4akp: like chanspy? |
| 17:02.00 | vk4akp | Yea like that. |
| 17:02.05 | vk4akp | I couldn't make it work though. |
| 17:02.25 | vk4akp | Mainly being able to monitor a meetme conference without chiming in. |
| 17:02.38 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
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| 17:03.37 | rocksfrow | is it possible to configure a polycom phone to ring forever based on specific route? |
| 17:04.14 | ManxPower | rocksfrow, route? |
| 17:04.23 | rocksfrow | what i meant by that was, cid i guess |
| 17:04.31 | rocksfrow | basically, i want the phone to ring forever if coming from the calling queue |
| 17:04.36 | rocksfrow | but ring for x seconds if anything else |
| 17:04.40 | rocksfrow | without using two extensions |
| 17:04.41 | rocksfrow | possible? |
| 17:04.48 | ManxPower | Figure out what you really want then ask again |
| 17:04.54 | rocksfrow | .... |
| 17:04.58 | rocksfrow | did what i just say not make sense |
| 17:05.15 | ManxPower | rocksfrow, set the polycom to ring (almost) forever, then do all your timeouts in the Asterisk dialplan |
| 17:05.39 | shader | or play ring tones using asterisk |
| 17:05.54 | shader | unless you mean timeout built into the phone |
| 17:06.00 | ManxPower | shader, that makes no sense whatsoever |
| 17:06.09 | rocksfrow | lol |
| 17:06.23 | rocksfrow | well, i have my calling queue setup to ring forever, instead of ringing / timing out / ringing / timing out |
| 17:06.35 | rocksfrow | but, the phones stop ringing after a bit, bc of the timeout on the phone i assume |
| 17:06.54 | rocksfrow | i was just curious if anybody knew a way to configure this directly within the phone |
| 17:07.21 | ManxPower | rocksfrow, YES! |
| 17:07.36 | ManxPower | The Polycom Admin Guide - the same place ALL information on Polycom setup is located. |
| 17:08.24 | BANSAL | Is there any difference using asterisk if I install it using fedora repo or if I install it using tar ball ? |
| 17:09.18 | *** join/#asterisk shader (~40846872@gateway/web/freenode/x-tcwhbtllrxqfruem) |
| 17:10.11 | BANSAL | actually I have install it from fedora repo .. and now I don't know how can I use additional module if I want ... |
| 17:10.13 | rocksfrow | ManxPower, lol, you're such a smartass |
| 17:10.23 | rocksfrow | nothing comes easy from you lol |
| 17:10.40 | rocksfrow | but thank you! |
| 17:13.00 | shader | rocksfrow: I'm somewhat confused as to why the phones would time out. Don't you have asterisk answer the connection before it puts them in a queue? |
| 17:13.33 | rocksfrow | shader, they're in the queue.. |
| 17:13.51 | rocksfrow | while they're in the queue, the agents are rang over and over |
| 17:13.58 | bmoraca_work | does anyone here happen to have a copy of the Cisco Unity Express Script Editor v2.3? |
| 17:14.09 | rocksfrow | if i configure it to timeout after 15 seconds and restart, it works fine... but at the same time leaves extra records in the history |
| 17:14.22 | rocksfrow | so, i reconfigured it to ring forever, but need the phones to also allow ringing forever |
| 17:14.44 | *** join/#asterisk DennisG (DennisG@2002:541e:88d0:0:2d71:78c6:a11f:183c) |
| 17:15.38 | shader | rocksfrow: so this is for the agents' phones? |
| 17:17.23 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 17:17.46 | shader | well, that does sound like something that would be on the phone itself |
| 17:18.00 | shader | unfortunately, there isn't a #polycom channel |
| 17:18.09 | Naikrovek | this is as close as you'll get |
| 17:18.20 | Naikrovek | there really isn't a need for #polycom with us in here :) |
| 17:18.32 | rocksfrow | Naikrovek, :-p |
| 17:18.51 | rocksfrow | im thinking about just making a second extension |
| 17:18.59 | shader | and yet, no one seems to want to answer rocksfrow's polycom specific questions |
| 17:19.19 | Naikrovek | we all have other things to do. most of us anyway |
| 17:19.19 | beefpastry | BANSAL: have to get the src rpm and rebuild if it's missing modules (in other words, might as well compile from source, anyway) |
| 17:19.31 | rocksfrow | Naikrovek, thanks, lol |
| 17:19.56 | Naikrovek | how long does the polycom ring before it stops |
| 17:20.42 | Naikrovek | ah i'm reading the backlog now |
| 17:20.45 | Naikrovek | one moment |
| 17:21.15 | Naikrovek | okay |
| 17:21.28 | [TK]D-Fender | [13:06]<rocksfrow>but, the phones stop ringing after a bit, bc of the timeout on the phone i assume <- Correct... this is an assumption |
| 17:21.40 | Naikrovek | what manxpower said is right, i think |
| 17:21.46 | rocksfrow | [TK]D-Fender, a pretty accurate one... |
| 17:21.55 | rocksfrow | [TK]D-Fender, i'm pretty positive, actually lol |
| 17:22.00 | rocksfrow | pretty common sense |
| 17:22.07 | Naikrovek | red flag |
| 17:22.11 | [TK]D-Fender | rocksfrow: People are usually very sure of things they aren't really looking at |
| 17:22.16 | rocksfrow | lol.. |
| 17:22.18 | rocksfrow | yes, i understand that |
| 17:22.26 | Naikrovek | whenever someone says something is common sense, red flag |
| 17:22.26 | rocksfrow | you have questions for me to confirm?.. |
| 17:22.30 | BANSAL | beefpastry: actually I am using fedora 12 and installed it from repo ... now for libpri and dahdi installation should I do all the work to /etc asterisk or anything else ? |
| 17:22.30 | rocksfrow | haha.. |
| 17:22.32 | rocksfrow | whatever |
| 17:22.36 | rocksfrow | shall i explain? |
| 17:22.46 | [TK]D-Fender | rocksfrow: Got a real call and real configs to show us? |
| 17:22.50 | Naikrovek | show us your config where you've set it to ring forever |
| 17:22.57 | Naikrovek | and show us the call log showing that it rings forever |
| 17:23.00 | rocksfrow | it's a freepbx system XO |
| 17:23.02 | rocksfrow | here it comes... |
| 17:23.04 | Naikrovek | yep |
| 17:23.04 | [TK]D-Fender | rocksfrow: I don't do "stories", I do "configs & debug" |
| 17:23.05 | rocksfrow | lol |
| 17:23.06 | Naikrovek | here it comes |
| 17:23.13 | BANSAL | beefpastry: /etc/asterisk |
| 17:23.44 | rocksfrow | this is a polycom specific question |
| 17:23.54 | rocksfrow | i'm simply asking how i can modify the ring timeout on the polycom |
| 17:23.56 | Naikrovek | then show us the polycom specific config files |
| 17:24.00 | Naikrovek | config files |
| 17:24.04 | Naikrovek | you modify config files |
| 17:24.09 | Naikrovek | show us your existing config files |
| 17:24.15 | rocksfrow | yeah..i was wondering if anybody knew the config off the top of their head |
| 17:24.20 | Naikrovek | jesus |
| 17:24.27 | Naikrovek | this is why you don't get an answer |
| 17:24.30 | rocksfrow | i'm provisioning via FTP thanks to somebodys help with some scripts here |
| 17:24.39 | Naikrovek | that might have been me |
| 17:24.41 | shader | [TK]D-Fender is really strict about that stuff. If you sneezed on him, he probably wouldn't give you a tissue until after you showed him the logs :D |
| 17:24.41 | rocksfrow | Naikrovek, ...are you serious? lol ...i'm bringing up the config damn |
| 17:24.42 | Naikrovek | show us the sip.cfg |
| 17:24.55 | [TK]D-Fender | 13:23]<rocksfrow>i'm simply asking how i can modify the ring timeout on the polycom <- there is no such thing |
| 17:25.14 | [TK]D-Fender | rocksfrow: You want to let a call ring pretty much forever? Go right ahead |
| 17:25.38 | Naikrovek | if [TK]D-Fender is right, as per usual, then its freepbx that is stopping the ringing |
| 17:25.44 | Naikrovek | so it ceases to become a polycom only question |
| 17:26.02 | rocksfrow | okay how about this |
| 17:26.07 | rocksfrow | i'll change it from unlimted to 3 minutes |
| 17:26.13 | rocksfrow | and watch the polycoms stop rining at the same exact time |
| 17:26.26 | [TK]D-Fender | rocksfrow: sure,... go show us |
| 17:32.09 | rocksfrow | yeah man.. |
| 17:32.53 | rocksfrow | so, i changed the retry timeout from unlimited, to 60 seconds, and the phone does as expected...it rings for a little less than 60 seconds..then restarts ringing a couple seconds after that |
| 17:33.02 | rocksfrow | because it restarts the ringing process.. |
| 17:33.14 | rocksfrow | i'm trying to set this to unlilmited so the phones don't go through that brief hesitation |
| 17:33.23 | rocksfrow | when i do that, the phones never start ringing again after the first cycle |
| 17:35.00 | *** join/#asterisk hfb (~hfb@pool-98-112-146-69.lsanca.dsl-w.verizon.net) |
| 17:37.04 | *** join/#asterisk b14ck (~b14ck@gibs-isp-fw-d02-v20.consolidated.net) |
| 17:41.21 | chazzam | rocksfrow: you still don't have logs or config... |
| 17:41.54 | rocksfrow | chazzam, bc im testing something |
| 17:42.01 | rocksfrow | #call.offeringTimeOut.label= |
| 17:42.01 | rocksfrow | call.offeringTimeOut.description=Time in seconds to allow an incoming call to ring before dropping the call, \ |
| 17:42.01 | rocksfrow | <PROTECTED> |
| 17:42.06 | rocksfrow | found that...^ |
| 17:42.17 | [TK]D-Fender | rocksfrow: And what was it set to? |
| 17:42.31 | rocksfrow | its set to 60, im making it 0 and reprovisioning to test |
| 17:43.40 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-143-123.dsl.stlsmo.sbcglobal.net) |
| 17:44.07 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 17:44.43 | LemensTS | I have ata/sip phones deployed around the USA off my asterisk server, im wanting to build a provision/firmware server. I have option of TFTP, HTTP, HTTPS on cisco SPA2102 what do you guys suggest I use |
| 17:45.11 | LemensTS | Im using Polycom Soundpoint phones also |
| 17:45.51 | LemensTS | Ive not read the provisioning docs on them, ive done a tftp server for polycom phones before...http and https im not sure on them |
| 17:45.53 | *** join/#asterisk QaDeS_ (~mklaus@p54A1B143.dip0.t-ipconnect.de) |
| 17:47.19 | Naikrovek | are http and https supported? yes |
| 17:47.43 | Naikrovek | ftp, tftp, ftps, http, https |
| 17:48.10 | LemensTS | What do you prefer? |
| 17:49.00 | Naikrovek | ftp |
| 17:49.02 | LemensTS | Guess ill have to have ftp/tftp for the firmware on both... |
| 17:49.17 | LemensTS | you like ftp for the .cfg also? |
| 17:49.20 | Naikrovek | yeah |
| 17:49.35 | Naikrovek | i put everything in a ftp directory, then tell the phones to log into that server |
| 17:49.44 | Naikrovek | everything goes in root directory from the phone's POV |
| 17:49.49 | Naikrovek | everything just works |
| 17:50.12 | LemensTS | Why not tftp? |
| 17:50.24 | Naikrovek | because i want to enforce passwords |
| 17:50.25 | LemensTS | TFTP seems finicky from what I have used.. |
| 17:50.36 | *** part/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 17:51.17 | *** join/#asterisk friartuck (~pmccary@66.162.90.56) |
| 17:51.57 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 17:52.07 | [TK]D-Fender | I don't treat my PBX as "trivial" |
| 17:57.43 | *** join/#asterisk BANSAL (~BANSAL@117.207.82.156) |
| 17:59.48 | leifmadsen | ok this is the first time I've really had an issue like this, so I'm curious if I should be trying something like "relaxdtmf" or the like. Problem is that I have a PRI on a Digium TE220p card. For some reason sometimes I get either duplicate DTMF on the first digit (sometimes) or it misses the first digit entirely. This seems to most often happen with '2' (oddly enough). It's not consistent, and I don't believe the card h |
| 17:59.48 | leifmadsen | as an echo cancel board on it (not 100% sure, researching that now). Configuration information here: http://pastebin.com/p1m2mcpV |
| 18:00.54 | rocksfrow | Naikrovek, do configurations overwrite if they're previously set..or no? CONFIG_FILES="preferences.cfg, ringer.cfg, server.cfg, 0004f22b1224-phone.cfg, phone1.cfg, sip.cfg" |
| 18:01.04 | leifmadsen | confirmed just now, no octasic echo cancel daughter card |
| 18:01.05 | rocksfrow | is a configuration in sip.cfg going to override something in xxxx-phone.cfg? |
| 18:02.03 | [TK]D-Fender | rocksfrow: So far you're drilling without looking at the simple proof. |
| 18:03.02 | *** join/#asterisk Canabinoide (~eu@189.96.54.38) |
| 18:03.29 | rocksfrow | [TK]D-Fender, i think you're right |
| 18:03.58 | *** join/#asterisk glwgoes (~guilherme@200.175.61.250) |
| 18:07.09 | BANSAL | hey guys I am having this problem installing dahdi .. http://pastebin.org/258654 |
| 18:07.18 | BANSAL | please take a look ... |
| 18:08.48 | [TK]D-Fender | BANSAL: You do not appear to have the sources for the 2.6.32.12-115.fc12.i686.PAE kernel installed. <- |
| 18:09.01 | [TK]D-Fender | BANSAL: Go install the pre-requisites that the docs tell you you need |
| 18:09.18 | BANSAL | [TK]D-Fender: I have install kernel-devel .. |
| 18:10.22 | BANSAL | but still the same problem .. |
| 18:10.29 | [TK]D-Fender | BANSAL: You need the devel, and the headers. Go prove precisely which of each you have installed. |
| 18:10.48 | BANSAL | [TK]D-Fender: yup I have install both ... |
| 18:11.04 | Slugs_ | he said prove |
| 18:12.27 | leifmadsen | BANSAL: you've installed the same version for the currently running kernel? |
| 18:12.30 | leifmadsen | likely not |
| 18:12.53 | BANSAL | http://pastebin.org/258682 |
| 18:12.55 | *** join/#asterisk k-man (~jason@unaffiliated/k-man) |
| 18:12.56 | leifmadsen | if you're running and older kernel than what is currently available, then the packages you install will be for the NEWEST kernel and not what you're running |
| 18:13.10 | leifmadsen | BANSAL: uname -a |
| 18:13.10 | BANSAL | leifmadsen: how can I check it ? |
| 18:13.18 | leifmadsen | what i running? |
| 18:13.23 | leifmadsen | what is running? |
| 18:13.34 | [TK]D-Fender | [14:08]<[TK]D-Fender>BANSAL: You do not appear to have the sources for the 2.6.32.12-115.fc12.i686.PAE kernel installed. <- |
| 18:13.36 | [TK]D-Fender | ^^^6 |
| 18:14.09 | [TK]D-Fender | BANSAL: None of those packages you show are for the PAE kernel |
| 18:14.11 | BANSAL | leifmadsen: Linux localhost.localdomain 2.6.32.12-115.fc12.i686.PAE #1 SMP Fri Apr 30 20:14:08 UTC 2010 i686 i686 i386 GNU/Linux |
| 18:14.19 | leifmadsen | what [TK]D-Fender said |
| 18:14.24 | leifmadsen | .PAE is your problem |
| 18:14.28 | leifmadsen | they do not matc h |
| 18:14.34 | BANSAL | [TK]D-Fender: which package .. can you tell ... |
| 18:14.42 | leifmadsen | both |
| 18:14.51 | leifmadsen | look at them - they are not the exact same string |
| 18:14.58 | leifmadsen | i686 != i686.PAE |
| 18:15.51 | BANSAL | <PROTECTED> |
| 18:16.08 | leifmadsen | BANSAL: the correct version of those packages |
| 18:16.59 | BANSAL | <PROTECTED> |
| 18:17.14 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:18.02 | pabelanger | BANSAL: rpm -q kernel-devel |
| 18:18.34 | BANSAL | kernel-devel-2.6.32.12-115.fc12.i686 |
| 18:18.35 | BANSAL | only this result ... |
| 18:19.32 | pabelanger | BANSAL: like leifmadsen said, you are running PAE kernel, but don't have source installed for it |
| 18:20.05 | BANSAL | <PROTECTED> |
| 18:20.08 | *** join/#asterisk dotnetted (~dotnetted@75.138.79.27) |
| 18:20.25 | BANSAL | this is not in repo I think ... |
| 18:20.44 | leifmadsen | you must be running a custom kernel or something -- google is likely helpful at this point |
| 18:21.06 | dotnetted | hey all - I just changed my asterisk users UID to 999 to hide it from the login choices on Ubuntu/Gnome and after a reboot asterisk seems to be missing all the sip commands - what might have happened? thanks |
| 18:21.26 | pabelanger | BANSAL: install the kernel headers. kernel-PAE-devel-2.6.32.12-115.fc12.i686.rpm |
| 18:21.43 | [TK]D-Fender | dotnetted: Did you change the owner on all of *'s files? |
| 18:21.54 | [TK]D-Fender | dotnetted: Because files actually store the NUMBER, not the name. |
| 18:22.13 | [TK]D-Fender | dotnetted: So if you just changed the mapping, the files remain with the old number and no name to match |
| 18:22.25 | pabelanger | BANSAL: http://lmgtfy.com/?q=fc12+pae+kernel-devel |
| 18:22.35 | dotnetted | oh doh - that would explain it heh - thanks |
| 18:22.54 | dotnetted | now to try to find all of the files... ;) |
| 18:23.24 | pabelanger | dotnetted: look at your asterisk.conf for the paths |
| 18:23.49 | dotnetted | pabelanger: thanks |
| 18:25.27 | *** join/#asterisk joako_ (~joako@opensuse/member/joak0) |
| 18:27.11 | dotnetted | after changing all the permissions whats the best way to reload asterisk (as it would on reboot) without rebooting? |
| 18:27.37 | shader | which dahdi package is required for meetme conferencing only: dahdi-linux, or dahdi-linux-complete? |
| 18:29.52 | *** join/#asterisk scalex000 (~chatzilla@190.166.188.12) |
| 18:29.57 | scalex000 | Hi guys |
| 18:30.21 | [TK]D-Fender | shader: The first includes al the drives. The latter includes toos you may never need regardless |
| 18:30.48 | *** join/#asterisk maddhat (~MadHatter@173-26-185-193.client.mchsi.com) |
| 18:31.21 | [TK]D-Fender | drivers* |
| 18:31.43 | jaytee | suspect TK's fingers are tired |
| 18:32.05 | maddhat | hi everyone. sorry for the newbie question... Wanting to setup an answering machine from my PSTN and i have 2x PCI modems laying around for it.. will either of these work? Gateway 6001761 Modem 17510, Compaq 56K Modem - CIS WS/M1-5614PM3 PM1560024001 |
| 18:32.31 | jaytee | I see a soldering gun and years of frustration in someone's future |
| 18:35.42 | Naikrovek | maddhat: won't work. modems don't do voice |
| 18:35.53 | jaytee | ~savemoney |
| 18:35.54 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 18:36.23 | pabelanger | jaytee: lol |
| 18:36.27 | pabelanger | nice macro |
| 18:36.35 | *** join/#asterisk dotnetted (~dotnetted@75.138.79.27) |
| 18:36.40 | jaytee | it's just infobot being a genius |
| 18:36.53 | *** join/#asterisk dohd (~Xaa@nala.dohd.org) |
| 18:36.54 | pabelanger | infobot: :) |
| 18:36.55 | infobot | (: |
| 18:37.09 | jaytee | ~botsnack |
| 18:37.09 | infobot | aw, gee, jaytee |
| 18:37.36 | pabelanger | jaytee: any way for infobot to list all of his commands? |
| 18:37.50 | jaytee | pabelanger, not sure really |
| 18:37.54 | dotnetted | whats the module called that provides sip functionality? |
| 18:38.04 | jaytee | chan_sip.so |
| 18:38.05 | Naikrovek | sip module? |
| 18:38.21 | BANSAL | pabelanger: I have installed dahdi ... but now there is error installing asterisk ... |
| 18:38.40 | pabelanger | infobot: what's up? |
| 18:38.40 | infobot | Up is the direction away from the central point of gravity. |
| 18:38.40 | BANSAL | pabelanger: make: *** [makeopts] Error 1 |
| 18:38.55 | pabelanger | ~pb |
| 18:38.56 | 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. |
| 18:38.56 | rocksfrow | is it possible to override this for a specific extension? |
| 18:38.59 | pabelanger | BANSAL: ^^ |
| 18:39.10 | BANSAL | yup .. sure .. |
| 18:39.32 | maddhat | Naikrovek: ok. darn :-/ well thank you for the information |
| 18:39.55 | Naikrovek | welcome |
| 18:40.11 | rocksfrow | Naikrovek, [TK]D-Fender im just blind.. |
| 18:40.14 | rocksfrow | and retarded |
| 18:40.21 | BANSAL | pabelanger: http://pastebin.org/258729 |
| 18:40.28 | Naikrovek | rocksfrow: what's the solution? |
| 18:40.46 | rocksfrow | was missing the ringtime setting specific to an extension |
| 18:40.57 | rocksfrow | my blindass was looking for 'timeout' |
| 18:40.58 | Naikrovek | gotcha |
| 18:41.01 | Naikrovek | heh |
| 18:41.03 | rocksfrow | not ringtime, heh |
| 18:41.07 | Naikrovek | freepbx setting, then |
| 18:41.16 | rocksfrow | hence my apologee :-) |
| 18:41.26 | rocksfrow | but there is a setting on the polycom phone too though :-) |
| 18:41.28 | rocksfrow | that sets to 60 |
| 18:41.34 | rocksfrow | so my bet is, when i jump this up..ill hit that limit |
| 18:44.12 | dotnetted | after changing asterisks UID I screwed up the permissions for many of the files it uses - I fixed all I could find but now "module load chan_sip.so" fails without much info - debugging is on and nothing useful was logged to log/messages or log/full - whats the best way to debug why the module failed to load? |
| 18:44.42 | [TK]D-Fender | dotnetted: Go prove the owners are right on everything |
| 18:45.00 | BANSAL | pabelanger: got anything ? |
| 18:46.00 | dotnetted | [TK]D-Fender: do you know which files would need to be set (at the very least) for chan_sip.so to load? I cant seem to find any that have the wrong permissions anywhere |
| 18:46.21 | [TK]D-Fender | dotnetted: Modules, configs... |
| 18:47.23 | dotnetted | - /usr/lib/asterisk/modules/* and /etc/asterisk/* all have the right permissions (and all the modules show up under 'module show') |
| 18:47.55 | [TK]D-Fender | dotnetted: better start showing us... |
| 18:48.24 | pabelanger | BANSAL: ./configure |
| 18:48.47 | BANSAL | pabelanger: done .. |
| 18:49.02 | pabelanger | BANSAL: make |
| 18:49.25 | BANSAL | pabelanger : after ./configure ... make ? |
| 18:49.42 | pabelanger | BANSAL: yes. read the INSTALL file |
| 18:49.51 | *** join/#asterisk Alagar (~Administr@122.164.38.169) |
| 18:49.53 | BANSAL | pabelanger: done .. same results |
| 18:50.08 | pabelanger | pastebin your config.log file |
| 18:50.12 | BANSAL | pabelanger: am I still missimng something ? |
| 18:51.35 | BANSAL | pabelanger: http://pastebin.org/258757 |
| 18:51.56 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:53.17 | pabelanger | BANSAL: you are missing your compilers, install them |
| 18:53.32 | BANSAL | as ... yum install ... |
| 18:54.13 | pabelanger | BANSAL: yes |
| 18:54.16 | pabelanger | BANSAL: http://lmgtfy.com/?q=asterisk+fedora+howto |
| 18:54.25 | pabelanger | follow the guides listed there |
| 18:57.32 | BANSAL | pabelanger: which compiler I have to install ? |
| 18:57.45 | BANSAL | pabelanger: I have installed gcc |
| 18:58.55 | pabelanger | BANSAL: config.log tells you what you are missing |
| 18:59.15 | pabelanger | BANSAL: http://www.asteriskguru.com/tutorials/asterisk_installation_compilation_fedora.html |
| 18:59.20 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 19:04.53 | *** join/#asterisk Z_God (~julius@wlan230085.mobiel.utwente.nl) |
| 19:08.13 | Qwell | looks at puzzled |
| 19:08.30 | Qwell | SMTP error from remote mail server after end of data: [83.163.53.136]: 550 5.7.1 No spam accepted |
| 19:08.57 | puzzled | hmm lemme check that. which To: ? |
| 19:09.12 | Qwell | patrick@ |
| 19:12.59 | puzzled | Qwell: please try again. I forgot to remove a temp block for some Chinese spam bot going crazy that used your name in From: |
| 19:13.11 | Qwell | seriously? |
| 19:13.31 | puzzled | only first name, not surname or first + surname |
| 19:13.34 | Qwell | oh |
| 19:14.42 | *** join/#asterisk knot (yiffstar66@unaffiliated/devemo) |
| 19:15.22 | puzzled | Qwell: got it. thanks for the response. will have a look |
| 19:16.20 | maddhat | Naikrovek: from what im reading here.. softmodems can be used: http://linuxgazette.net/120/smith.html |
| 19:16.35 | maddhat | there something im missing? |
| 19:17.21 | leifmadsen | maddhat: not any soft modem |
| 19:17.23 | Qwell | maddhat: That specific one. but it's crap. |
| 19:17.31 | leifmadsen | super crap |
| 19:17.36 | maddhat | i see |
| 19:17.47 | leifmadsen | you're way better off with an FXO -> SIP device |
| 19:17.48 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:18.07 | idespinner | spa3102 is only 60$ |
| 19:18.09 | maddhat | but looking at the price of the asterisk cards.. yikes! |
| 19:18.16 | idespinner | 1fxo 1 fxs |
| 19:20.16 | Qwell | maddhat: if that scares you, go look at how much "real" PBX hardware costs. |
| 19:21.09 | maddhat | im not arguing that its an extremely low priced alternative.. but for a hobbyist like me who just wants simple features it seems like overkill |
| 19:25.18 | Corydon76-dig | maddhat: it's actually quite reasonable for hardware |
| 19:25.29 | pabelanger | maddhat: TDM400P is sub $200 |
| 19:25.41 | pabelanger | depending on how many ports you get |
| 19:25.48 | Corydon76-dig | TDM410 is what he wants |
| 19:25.54 | pabelanger | SPA3102 is another option |
| 19:25.56 | maddhat | what about just something like this: http://cgi.ebay.com/X100PSE-FXO-PCI-Digium-Asterisk-Trixbox-Elastix-FreePBX-/130339643798?cmd=ViewItem&pt=LH_DefaultDomain_0&hash=item1e58d92196 |
| 19:26.52 | Corydon76-dig | maddhat: I have one of those cards. Trust me. You want the TDM410 card |
| 19:27.18 | [TK]D-Fender | madWhat are you looking to do exactly? |
| 19:27.22 | pabelanger | Is the X100p even supported any more? |
| 19:27.36 | Corydon76-dig | pabelanger: we've never removed support for it in DAHDI |
| 19:27.37 | [TK]D-Fender | maddhat: What are you looking to do exactly? |
| 19:27.57 | pabelanger | Corydon76-dig: gotcha |
| 19:29.10 | maddhat | [TK]D-Fender: id like to setup a simple voicemail box... and possibly look into getting a SIP phone |
| 19:29.45 | [TK]D-Fender | maddhat: For your single analog line? |
| 19:29.49 | maddhat | correct |
| 19:31.15 | Corydon76-dig | maddhat: if you can accept a card that works to a degree, but I don't think will work to your complete satisfaction, then get the X100P |
| 19:31.45 | lirakis | is there a config option to tell asterisk where sounds reside? |
| 19:31.50 | Corydon76-dig | But don't get it if you think it's going to be the last telephony card you'll ever need |
| 19:31.55 | *** join/#asterisk shader (~40846872@gateway/web/freenode/x-bwojznmwmbgntksw) |
| 19:31.58 | maddhat | is voice quality the difference between the two? |
| 19:32.20 | Corydon76-dig | maddhat: and echo, callerid success... |
| 19:32.21 | [TK]D-Fender | maddhat: The SPA-3102 may be a better bet for you. You'll get the FXO side out of the way, reduce system requirements, and get an FXS interface to use an analog hone with your system immediately. |
| 19:32.26 | shader | any good references for providing IMAP access to asterisk voicemail? |
| 19:33.46 | lirakis | astvarlibdir ? |
| 19:33.58 | Corydon76-dig | [TK]D-Fender: for someone who's not yet familiar with Asterisk, I'm not sure I'd recommend an external FXO device |
| 19:34.33 | shader | nvm, I found a section on it in The Book |
| 19:34.42 | Corydon76-dig | Most of the time, people want to have a low-effort audio path, then they can branch out |
| 19:34.48 | pabelanger | lirakis: yes |
| 19:34.58 | [TK]D-Fender | Corydon76-dig: Compare filling in a few small blanks VS the giant story we got from this other guy a few minutes ago who can't compile shit, and all those with bitchy kernel, ones who get updaetd all the time , etc. |
| 19:35.08 | [TK]D-Fender | Corydon76-dig: Seriously... almost a non-issue |
| 19:35.28 | lirakis | hrm ... asterisk is saying it cant open tt-monkeys (ulaw) ... but its def. there, and the sound file directory is correct |
| 19:35.46 | maddhat | i enjoy tinkering... and am no stranger to networking configuration.. im sure i can manage setup |
| 19:35.52 | [TK]D-Fender | Corydon76-dig: there are certain things that a DAHDI driven card can do better for sure... but in many other cases I figure it isn't worth it for the "el-cheapo" side |
| 19:36.00 | lirakis | and ... i have ulaw sounds... and that is what my codec has been negotiated to... i wonder if there is some i/o prob. |
| 19:37.08 | pabelanger | lirakis: pastebin output |
| 19:37.47 | maddhat | thanks for the options everyone. ill look into the 3102 more seriously as an option. |
| 19:37.59 | lirakis | pabelanger, http://pastie.org/969987 |
| 19:39.26 | pabelanger | Linuturk: ls /var/lib/asterisk/sounds/en/ | grep tt-monkeys |
| 19:39.31 | *** join/#asterisk italorossi (~italoross@201.76.154.130.intranet.digi.com.br) |
| 19:40.08 | lirakis | pabelanger, http://pastie.org/969992 |
| 19:40.55 | lirakis | pabelanger, ... asterisk.conf ... http://pastie.org/969995 |
| 19:41.13 | lirakis | pabelanger, my sounds are not in the en folder |
| 19:41.17 | lirakis | ... do they need to be? |
| 19:41.22 | pabelanger | lirakis: yes |
| 19:41.25 | lirakis | i guess if i have the directory prefix thingy set |
| 19:41.28 | pabelanger | if that is your default language |
| 19:41.28 | lirakis | ok .. ill move |
| 19:41.35 | italorossi | Hello all, I'm getting this message: "audiohook_read_frame_both: Read factory 0xa27c300 was pretty quick last time, waiting for them.". Does anyone know the cause of this? I've tested with different asterisk 1.4 versions (26, 28, 31). |
| 19:42.34 | file | that message is primarily for debugging ... |
| 19:42.40 | *** part/#asterisk maddhat (~MadHatter@173-26-185-193.client.mchsi.com) |
| 19:44.24 | lirakis | pabelanger, do i need to set language=en in sip.conf ?? |
| 19:44.56 | italorossi | ok.. this can cause some strange behaviors in recorded calls? Calls with silence or repeating the audio at the end? |
| 19:45.06 | pabelanger | lirakis: no, it is the default |
| 19:45.53 | lirakis | hmmm |
| 19:45.53 | file | italorossi, it won't cause strange behaviors - it's a perfectly normal message to have under normal circumstances |
| 19:45.53 | lirakis | pabelanger, still getting the same thi8ng |
| 19:46.30 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 19:48.11 | lirakis | i think there is an i/o issue ... this was working with voicemail before |
| 19:49.31 | *** join/#asterisk david456 (~david7345@118.172.90.13.adsl.dynamic.totbb.net) |
| 19:49.42 | lirakis | and now i get the same crap for voicemail ... saying that it the vm sound files dont exist |
| 19:50.03 | muiro | Does the 'allowmultiplelogin' setting in manager.conf disallow multiple logins based on the manager user ID or the host/ip? |
| 19:50.07 | pabelanger | lirakis: what did you change on your system, if this used to work |
| 19:50.36 | lirakis | pabelanger: i changed what you just suggested .. moved files to the en directory |
| 19:50.45 | italorossi | file: thanks! I'm having some problems with call recordings and this message called my atention. I have the same configuration (boards, asterisk version, call flow) in other client and I don't see the same behavior. |
| 19:51.26 | lirakis | pabelanger, this is running on a hacked wrt54g router.... i put a 2 gig sd card in running serial i/o over free gpio pins on the cpu .... so.... my guess is there is some issue with reading |
| 19:52.06 | lirakis | is waiting for my fanless 1u server to show up so i can stop running on 13mb of ram |
| 19:52.07 | muiro | Every time I set 'allowmultiplelogin' to 'no', no manager is able to connect to the system at all. Not even the first one |
| 19:52.21 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.7 (2010/05/04), 1.6.0.28, 1.6.1.20 (2010/05/20), 1.4.31 (2010/05/04), *-Addons 1.6.2.1, 1.6.1.3, 1.6.0.5, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 19:52.48 | pabelanger | lirakis: maybe, then move your files back to the previous directory for VM to work again |
| 19:53.03 | leifmadsen | Asterisk 1.6.1.20 and 1.6.0.28 are now available. These are the final maintenance releases for the 1.6.0 and 1.6.1 branch. Please see the release announcement: http://www.asterisk.org/node/51358 |
| 19:53.28 | Naikrovek | final? call them 1.6.0.99 and 1.6.1.99 |
| 19:53.40 | Naikrovek | kidding |
| 19:53.43 | Naikrovek | i know it doesn't work that way |
| 19:53.59 | pabelanger | muiro: There must already be a manager login used. ;IF set to no, rejects manager logins that are already in use. |
| 19:54.00 | sprite-- | Has anyone ever experienced syslog.cron killing asterisk? http://gist.github.com/407996 |
| 19:55.15 | *** join/#asterisk sprite-- (~sprite@c-98-251-108-29.hsd1.ga.comcast.net) |
| 19:55.18 | sprite-- | sorry got disconnected |
| 19:56.36 | muiro | pabelanger: the very first manager login that I attempt fails on could not authenticate. If I set allowmultiplelogins to yes, it does not fail. I even check 'manager show connected' before trying this and no use is connected. |
| 19:56.46 | ryanlin | rejected because extension not found. |
| 19:57.11 | ryanlin | does anyone know why? |
| 19:57.28 | ryanlin | this is coming from our callmanager to asteriks |
| 19:57.30 | ryanlin | asterisk |
| 19:57.36 | [TK]D-Fender | ryanlin: Because there is no extension to match hat was dialed in the context the call is looking in <- |
| 19:57.40 | [TK]D-Fender | ryanlin: Just like it says |
| 19:58.06 | ryanlin | interesting |
| 19:58.15 | ryanlin | exten => 3300,1,Macro(corpuser,3300) ;test-phone |
| 19:58.21 | ryanlin | i have this entry defined in extensions.conf |
| 19:58.32 | ryanlin | and the appropiate entry defined in sip.conf for the actual phone config |
| 19:58.53 | ryanlin | this should be sufficient..i don't know why it's complaining that the extension is not found |
| 19:58.55 | [TK]D-Fender | ryanlin: Nowhere do I see a COMPLETE call with SIP DEBUG showing what CONTEXT its looking in, nor enough dialplan to prove that that is the one its looking for, and that its in the right place |
| 19:59.01 | [TK]D-Fender | ryanlin: PASTEBIN is your friend. |
| 19:59.03 | [TK]D-Fender | ~pb |
| 19:59.03 | 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. |
| 19:59.04 | ryanlin | one sec |
| 19:59.04 | [TK]D-Fender | ^^^^^^^^ |
| 19:59.12 | ryanlin | hold |
| 20:01.14 | ryanlin | http://internetworkpro.org/pastebin/6386 |
| 20:01.38 | ryanlin | the initial invite looks okay |
| 20:02.28 | pabelanger | ryanlin: what context did you setup in your sip.conf? |
| 20:02.33 | david456 | Hello, friends. Is there a preference order that asterisk has for sound file formats? Or does it just depend on the order in which the modules loaded? |
| 20:02.39 | [TK]D-Fender | ryanlin: Where is your dialplan to match? |
| 20:03.03 | [TK]D-Fender | pabelanger: We already see that answer |
| 20:03.07 | shader | for only meetme conferencing, which dahdi package do I need, dahdi-linux-complete, or just dahdi-linux? |
| 20:03.18 | [TK]D-Fender | shader: Latter. I already answered this |
| 20:03.27 | shader | yeah, I think I disconned for some reason |
| 20:03.38 | shader | so I didn't get to hear, sorry |
| 20:04.11 | shader | will I need dahdi-tools? |
| 20:04.52 | [TK]D-Fender | shader: Probably not |
| 20:05.36 | italorossi | Setting the AUDIOHOOK_INHERIT before or after the MixMonitor makes any difference? |
| 20:05.53 | shader | ok, thanks |
| 20:06.41 | pabelanger | david456: it all depends on what format you are providing to your phones. |
| 20:07.19 | pabelanger | david456: if your phone need ulaw, then install .ulaw. You don't want to be transcoding audio |
| 20:07.19 | sprite-- | http://gist.github.com/408010 why the hell does sysklogd restart my asterisk? |
| 20:07.25 | Trixboxer | Hi |
| 20:07.46 | Trixboxer | how can I dial a SIP URI from my dialpla ?? any idea ? |
| 20:07.51 | Trixboxer | dialplan* |
| 20:08.11 | pabelanger | sprite--: Because you told syslogd to restart it. check your syslogd scripts |
| 20:08.35 | pabelanger | ~book |
| 20:08.36 | 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 |
| 20:08.40 | pabelanger | Trixboxer: ^^^ |
| 20:09.16 | Trixboxer | thnx pabelanger.. will have a look at it... |
| 20:10.01 | ryanlin | one sec |
| 20:11.25 | *** join/#asterisk sahafeez (~sahafeez@65-119-47-100.dia.static.qwest.net) |
| 20:15.50 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 20:16.00 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 20:19.00 | *** join/#asterisk bent_screwdriver (~socain00@74.255.249.66) |
| 20:19.21 | bent_screwdriver | is there a way to change which wav format MixMonitor uses so I can use the wav format that makes for a smaller file? |
| 20:20.12 | [TK]D-Fender | bent_screwdriver: It records in precisely the format you specify |
| 20:21.37 | bent_screwdriver | [TK]D-Fender: looked on http://www.voip-info.org/wiki/view/Asterisk+cmd+MixMonitor and didn't see that opotion. Can you give me an example syntax? |
| 20:22.39 | [TK]D-Fender | bent_screwdriver: the file extension specifies the format |
| 20:24.13 | ryanlin | [TK]D-Fender: are you therE |
| 20:24.13 | bent_screwdriver | [TK]D-Fender: ahhh so to get wav49 do i do .wav or .WAV? |
| 20:24.40 | bent_screwdriver | ryanlin: TK's always here as far as I can tell ;) |
| 20:24.53 | [TK]D-Fender | ryanlin: Yes.. now where is the pastebin I asked for over 20 mins ago? |
| 20:25.04 | [TK]D-Fender | bent_screwdriver: Good for noticing that WAV != wav |
| 20:25.12 | ryanlin | [TK]D-Fender: http://internetworkpro.org/pastebin/6387 |
| 20:25.23 | [TK]D-Fender | bent_screwdriver: Feel free to read up on which is which, or do the 10 second test youself |
| 20:25.40 | [TK]D-Fender | ryanlin: that is NOT your dialplan |
| 20:26.03 | bent_screwdriver | [TK]D-Fender: will do. thanks for the info! |
| 20:26.13 | ryanlin | sorry |
| 20:26.14 | ryanlin | http://internetworkpro.org/pastebin/6388 |
| 20:26.16 | ryanlin | here it is |
| 20:26.44 | [TK]D-Fender | ryanlin: WTF is that? |
| 20:26.49 | ryanlin | isn't that the one? |
| 20:26.54 | [TK]D-Fender | ryanlin: That doesn't look like a proper extensions.conf |
| 20:27.00 | ryanlin | ahhh |
| 20:27.07 | ryanlin | shit..okay let me get all the stuff in extensions.conf |
| 20:27.08 | [TK]D-Fender | default => 3258,1,Dial(SIP/$) <-- PARDON? |
| 20:27.22 | [TK]D-Fender | ryanlin: Where did you just pull that mess from? |
| 20:27.30 | ryanlin | in extensions.conf |
| 20:27.34 | ryanlin | it was there |
| 20:27.43 | ryanlin | i commented it out |
| 20:27.52 | *** join/#asterisk boch (~boch@200.61.191.9) |
| 20:27.57 | [TK]D-Fender | ryanlin: I don't see contexts with braces around them, I see a single mashed line with junk in it. |
| 20:28.01 | [TK]D-Fender | RyTry again. |
| 20:29.02 | ryanlin | hmmm..in this case, i did not specify any dial plans for the extension 3300 |
| 20:29.06 | boch | Hi all, im having a problem, i have a DID point to my asterisk and the exten for that DID is configured OK, the fact is sometimes goes perfect but sometimes the call is rejected because exten was not found. never had a problem like this, what may i be doing wrong ? |
| 20:29.13 | ryanlin | or unless there is a default dial plan |
| 20:29.20 | [TK]D-Fender | ryanlin: No such thing. |
| 20:29.25 | ryanlin | ok |
| 20:29.29 | ryanlin | that explains it |
| 20:29.34 | [TK]D-Fender | ryanlin: And you are reversing things again. |
| 20:29.41 | [TK]D-Fender | An extension IS dialplan. |
| 20:29.55 | ryanlin | exten => 3300,1,Macro(corpuser,3300) ;test-phone |
| 20:29.58 | ryanlin | this is it then ? |
| 20:30.05 | ryanlin | this is what's in the extensions.conf |
| 20:30.11 | [TK]D-Fender | ryanlin: That is an extension. The question is.. WHERE is it? |
| 20:30.18 | ryanlin | in extensions.conf |
| 20:30.27 | [TK]D-Fender | WHAT FUCKING **CONTEXT** |
| 20:30.33 | [TK]D-Fender | bursts a vein |
| 20:30.56 | ryanlin | hmm |
| 20:31.10 | ryanlin | it's probably not defined then |
| 20:31.12 | [TK]D-Fender | ryanlin: pastebin the entire file. |
| 20:31.15 | ryanlin | sure |
| 20:31.22 | [TK]D-Fender | Ry"probably"? Why is this a guess? |
| 20:31.23 | *** join/#asterisk xheliox (~jeff@i216-58-41-253.cybersurf.com) |
| 20:35.56 | [TK]D-Fender | Ok, times up... |
| 20:35.58 | [TK]D-Fender | checkout time. |
| 20:36.00 | [TK]D-Fender | BBL |
| 20:43.27 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 20:47.08 | *** join/#asterisk Milad (~milad@unaffiliated/slackark) |
| 20:53.09 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 21:06.44 | *** join/#asterisk bmg505 (~leon@196-209-7-58-rndf-esr-3.dynamic.isadsl.co.za) |
| 21:17.19 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 21:17.46 | ruben23 | hi there any sofphones used iax2 with g729 codec..? |
| 21:18.41 | ruben23 | i would like to used for remote callers |
| 21:23.24 | *** join/#asterisk AndreBrasil (~staastis@187.65.196.165) |
| 21:25.19 | leifmadsen | ruben23: zoiper likely |
| 21:25.20 | ruben23 | anyone have idea..? |
| 21:25.29 | leifmadsen | patience grasshopper |
| 21:25.48 | ruben23 | leifmadsen:ok sorry, yeah zoiper free dont have g729 |
| 21:25.59 | leifmadsen | ruben23: that's because g729 is not a free codec |
| 21:26.17 | leifmadsen | it requires a license to use it |
| 21:26.37 | leifmadsen | you won't find a free softphone with g.729 support |
| 21:26.54 | Corydon76-dig | until 2014, when the associated patents expire |
| 21:26.55 | LemensTS | where do you get firmware for PAP2T? All I can find is the spc tool |
| 21:27.18 | ruben23 | <PROTECTED> |
| 21:27.31 | leifmadsen | ruben23: define "best" |
| 21:27.41 | Corydon76-dig | ruben23: lpc10 |
| 21:27.44 | ruben23 | currently im suing ulaw/alaw but getting bad connection and call drop |
| 21:27.55 | Corydon76-dig | ruben23: It's actually even better than g729 |
| 21:28.05 | leifmadsen | snickers |
| 21:28.17 | ruben23 | <PROTECTED> |
| 21:28.19 | *** part/#asterisk LemensTS (~LemensTS@adsl-70-238-143-123.dsl.stlsmo.sbcglobal.net) |
| 21:28.35 | leifmadsen | ruben23: potentially gsm I guess. Quality is not as good though. |
| 21:28.43 | *** join/#asterisk mboehn (mathias@mboehn.alfa-skk.pr0jectX.net) |
| 21:28.52 | leifmadsen | there is a reason lots of people want to use g.729 instead of a free codec |
| 21:28.58 | ruben23 | Corydon76-dig:its added by default on asterisk and most of the softphones right..? |
| 21:29.16 | leifmadsen | ruben23: he's yanking your chain -- lpc10 makes you sound like a robot |
| 21:29.20 | Corydon76-dig | I believe so, yes |
| 21:29.32 | Corydon76-dig | leifmadsen: it's the best codec, though... |
| 21:29.39 | leifmadsen | it certainly makes me laugh :) |
| 21:30.54 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 21:31.21 | shader | leifmadsen: is the stuff in the section in your book on imap voice mail really all you have to do to get imap working? |
| 21:31.39 | leifmadsen | shader: when we wrote it, yes |
| 21:31.47 | leifmadsen | I haven't done a lot of IMAP stuff and we actually had someone else write that |
| 21:32.05 | shader | so, there's no need to do any more configuration on dovecot? |
| 21:32.17 | leifmadsen | you ask a question I do not know the answer to |
| 21:32.23 | shader | ok |
| 21:32.43 | leifmadsen | check the imap documentation in the doc/ directory of your asterisk source |
| 21:32.49 | leifmadsen | it contains some more up to date information |
| 21:34.07 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:38.34 | italorossi | It's ok to define twice AUDIOHOOK_INHERIT in this case: Caller dial 7089 and 7089 executes an attended transfer to 7099 ? Example: http://pastebin.com/R0xGv6Tm |
| 21:39.04 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:41.41 | scalex000 | hello guys, I need to handle incoming call, but I have dude using queue. I don't if I need to dial first and when all represantive is busy to queue a call what is the suggestion |
| 21:42.11 | leifmadsen | GotoIf($[${DIALSTATUS} = BUSY]?send_to_queue) |
| 21:42.31 | *** join/#asterisk codefreeze-lap (~murf@mail.parsetree.com) |
| 21:42.34 | leifmadsen | if I at all understand what you just said |
| 21:42.55 | *** join/#asterisk nargon (mike@217.194.139.22) |
| 21:43.03 | [TK]D-Fender | leifmadsen: Yeah, that's mostly what he said... but likely not what he'll need :) |
| 21:43.06 | scalex000 | check this |
| 21:43.11 | leifmadsen | [TK]D-Fender: probably |
| 21:43.15 | [TK]D-Fender | leifmadsen: Which won't be any fault of yours naturally ;) |
| 21:43.18 | leifmadsen | goes back to doing work |
| 21:44.00 | *** part/#asterisk Canabinoide (~eu@189.96.54.38) |
| 21:44.12 | scalex000 | leifmadsen, http://pastebin.com/8WGCUnnJ |
| 21:44.12 | nargon | anyone know what to do, bv keeps disconnecting me after a few mnts.. i've changed qualify to = 3600.. i'm blocked right now on all the serves.. very annoying |
| 21:45.19 | *** join/#asterisk devmod (~devmod__@c-76-100-208-204.hsd1.md.comcast.net) |
| 21:45.27 | [TK]D-Fender | scalex000: thats kinda retarded... Starting MixMonitor and immediately hanging up the call... |
| 21:46.16 | scalex000 | TK, sorry my mistake |
| 21:46.31 | scalex000 | TK, anyway my confusion is over Dial |
| 21:46.47 | italorossi | any comments about this: http://pastebin.com/R0xGv6Tm ? |
| 21:46.53 | scalex000 | TK, how asterisk run gotoif |
| 21:47.23 | [TK]D-Fender | scalex000: Rephrase. You aren't making any sense? |
| 21:47.32 | [TK]D-Fender | s/?/./ |
| 21:47.37 | scalex000 | ok |
| 21:47.54 | nargon | italorossi is line 6 supposed to be n or 1 ? |
| 21:48.12 | *** join/#asterisk x303 (~X303@97.102.28.28) |
| 21:48.33 | italorossi | n, sorry! |
| 21:48.50 | italorossi | It's ok to define twice AUDIOHOOK_INHERIT in this case: Caller dial 7089 and 7089 executes an attended transfer to 7099 ? |
| 21:49.19 | scalex000 | TK, I mean I want before agent pickup the phone playback, active mixmonitor so when all are busy queue |
| 21:50.24 | scalex000 | TK, I think I found the answer, I need to dial to Agent not to SIP extension, what do you think? remember my english is poor |
| 21:50.34 | [TK]D-Fender | scalex000: Just look at the order you are doing things in. Clearly this isn't doing what you want. Go change it |
| 21:50.35 | leifmadsen | we noticed :) |
| 21:50.51 | scalex000 | leifmadsen, thx |
| 21:51.06 | [TK]D-Fender | scalex000: You should be starting Monitor before some other kind of call. |
| 21:51.36 | scalex000 | ok |
| 21:51.51 | *** join/#asterisk Get_The_Fish (~Get_The_F@173-14-4-113-Colorado.hfc.comcastbusiness.net) |
| 21:52.41 | nargon | can anyone comment registering a peer to broadvoice for incoming? the peer registeres somtimes but then goes to unregistered after a couple minutes then the server blocks my ast box for an unkown amount of time... i see posts on this on the net but i can't figure out a fix |
| 21:53.16 | *** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no) |
| 21:54.47 | *** join/#asterisk saisoma (~saisoma@client105.jdcc.edu) |
| 21:55.12 | [TK]D-Fender | nargon: you should pastebin the SIP debug of your actual registration attempts... |
| 21:55.16 | [TK]D-Fender | ~pb |
| 21:55.17 | 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. |
| 21:55.56 | scalex000 | TK, fix it http://pastebin.com/d4rRwqyB |
| 21:57.00 | nargon | [TK]D-Fender yeah its not really a problem registering its that the broadvoice servers block my registration when I register more than once every 60-120 seconds... occasionally they will accept my registration and then drop it after a few mnts... so all you would really see would be repeating reg attempts.. its not a problem with the config other than the picky bv registration crap.. |
| 21:57.44 | [TK]D-Fender | scalex000: Fix what? You haven't expressed what is FAILING there, |
| 21:57.58 | [TK]D-Fender | scalex000: And your GotoIF is bad. You have quotes on ONE SIDE only. |
| 21:58.14 | [TK]D-Fender | scalex000: the left side of your expression will never have quotes in the reulst |
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| 21:59.15 | nargon | 2142421756/2142421756 147.135.8.128 5060 OK (756 ms) registered now... will see how long it stays that way |
| 21:59.16 | scalex000 | TK, yes its terrible |
| 21:59.47 | nargon | don't laugh at my latency i'm on a satellite connection :) |
| 22:00.42 | [TK]D-Fender | result* |
| 22:00.43 | codefreeze-lap | LOL.... oops! Sorry! (murf shuts back up) |
| 22:00.50 | nargon | 2142421756/2142421756 147.135.8.128 5060 UNREACHABLE << yeah its dead now again |
| 22:01.35 | *** join/#asterisk trelane (~trelane@funtoo/staff/trelane) |
| 22:01.38 | [TK]D-Fender | nargon: that is only your peer status, it has nothing to do with your registration. |
| 22:01.56 | ManxPower | What is funny is that your users apparently tolerate 756 ms of latency. |
| 22:02.01 | trelane | need a recommendation on a single port FXO (receiving an inbound POTS line) that registered back to asterisk on the back end |
| 22:02.08 | ManxPower | "sip show registry" is what you want, I think. |
| 22:02.20 | trelane | (registers back via IAX or SIP) |
| 22:02.30 | [TK]D-Fender | trelane: SPA-3102 |
| 22:03.41 | nargon | ManxPower yeah show sip registry is fine.. it shows the status... not much i can get from that |
| 22:03.57 | trelane | [TK]D-Fender, config's similar to a linksys phone or a pap-2002? |
| 22:04.14 | nargon | trelane yeah we are in iraq.. actually not that bad.. somtimes i call people and they tell me its the clearest call they have every been on and I sound like i'm right down the street.. |
| 22:04.42 | trelane | nargon, cool |
| 22:05.58 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
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| 22:06.31 | [TK]D-Fender | trelane: Similar |
| 22:07.28 | trelane | [TK]D-Fender, many thanks |
| 22:07.33 | trelane | the FXS on that actually helps as well |
| 22:07.35 | *** join/#asterisk Mw3_ (mw3@mw3.hu) |
| 22:07.39 | trelane | oddly useful little device |
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| 22:22.39 | bcrisp | book |
| 22:23.13 | bcrisp | could someone drop a link to the free * ebook into chat? |
| 22:23.43 | *** join/#asterisk blaines (~blaines@75-171-72-110.phnx.qwest.net) |
| 22:25.50 | idespinner | you mean this: http://www.asteriskdocs.org/? |
| 22:26.16 | bcrisp | nah, got it but thanks |
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| 22:28.41 | ManxPower | ~answers |
| 22:28.42 | infobot | [~answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & Docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: /path/to/src/asterisk/UPGRADE*.txt |
| 22:28.55 | bcrisp | ~book |
| 22:28.56 | 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 |
| 22:29.03 | bcrisp | ok i forgot my tilde |
| 22:29.17 | nargon | how can i set my registration timeout from 120 to 3600 on my peer to bv ? |
| 22:29.51 | nargon | defaultexpirey=1800 |
| 22:29.52 | ManxPower | nargon, remove the change you made to the default of 3600 seconds? |
| 22:29.53 | nargon | maxexpirey=3600 |
| 22:30.00 | bcrisp | whats the default digittimeout? |
| 22:30.07 | nargon | ManxPower that was for qualify |
| 22:30.20 | nargon | should i just set default and max to 3600 ? |
| 22:30.20 | ManxPower | bcrisp, no idea, but ONLY applies to IVRs in Asterisk, not dialing |
| 22:30.21 | Corydon76-dig | bcrisp: 5 |
| 22:30.30 | bcrisp | Corydon76-dig: thanks |
| 22:30.36 | ManxPower | nargon, no, qualify defaults to something tless than 3600 |
| 22:30.56 | nargon | ManxPower i mean i put qualify=3600 |
| 22:31.03 | nargon | before i never modified the registration timeout |
| 22:31.32 | Corydon76-dig | qualify defaults to 2000 |
| 22:32.07 | Corydon76-dig | 3600 tells me you probably think it's doing something other than what you think |
| 22:32.08 | ManxPower | nargon, next time try SAYING "qualify" and not "registration" |
| 22:32.37 | Corydon76-dig | qualify == number of milliseconds a response needs to arrive before it's considered too late. |
| 22:32.53 | nargon | ManxPower i did :( |
| 22:33.02 | ManxPower | <nargon> how can i set my registration timeout from 120 to 3600 on my peer to bv ? |
| 22:33.11 | ManxPower | Then you started talking about qualify |
| 22:33.31 | ManxPower | Tell us again what you are trying to do. |
| 22:33.33 | nargon | right because you refered to an earlier comment where i was talking about qualify |
| 22:33.48 | ManxPower | nargon, no, I wasn't. |
| 22:33.57 | nargon | ok sorry thought you were |
| 22:35.09 | nargon | Corydon76-dig my default reg seems to have this line in the sip debug Expires: 120 i added maxexpirey and defaultexpiry = 3600 let me check debug again |
| 22:35.43 | ManxPower | nargon, I suspect BV is telling you to use 120 |
| 22:36.54 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 22:37.16 | nargon | ManxPower BV has a problem with you registering to often at 60 or 120 thats why i'm trying to set it to 3600 |
| 22:37.40 | nargon | its part of the kind of picky system they use to protect their servers i guess... |
| 22:38.13 | nargon | after you reg to often they drop all reg attempts for an seemingly random amount of time (this is what i'm getting of google) |
| 22:38.50 | Corydon76-dig | nargon: the default expiry for your registration is specified in your 'register' line, right after the '~' |
| 22:38.55 | nargon | http://www.voip-info.org/wiki/view/Broadvoice |
| 22:39.15 | nargon | ok let me try to look at that |
| 22:39.20 | ruben23 | hi i have an asterisk server whihc uses 10 Mbps this is shared betwwen data and voice, and got 13 phone extensions only, problem with calls are sometimes choppy voice and when someone call they cant hear each other. |
| 22:39.37 | ruben23 | this asterisk server is public and not in a NAT environment. |
| 22:40.05 | *** join/#asterisk knot (yiffstar66@unaffiliated/devemo) |
| 22:40.06 | ruben23 | si this caused by the data traffic..? but i have 10 Mb bandwidth. |
| 22:40.09 | Corydon76-dig | nargon: those other values are for regulating INCOMING registration requests from phones, not OUTGOING |
| 22:41.05 | nargon | Corydon76-dig right now i'm just trying to get the incoming peer to stay registered |
| 22:41.53 | nargon | ruben23 you can check your jitter by looking at stdev on a ping -c 100 yourhost.com |
| 22:41.56 | Corydon76-dig | nargon: No, if you're dealing with broadvoice, you're working with the OUTGOING peer |
| 22:42.42 | nargon | Corydon76-dig in trixbox I have a user set up for outgoing and a peer set up for incoming |
| 22:43.05 | Corydon76-dig | nargon: there's your overarching problem. IT's trixbox |
| 22:43.12 | Corydon76-dig | ~trixbox |
| 22:43.13 | infobot | i heard trixbox is SH1TB0X. Basically a CRAPPY, closed-source distro. STAY AWAY! |
| 22:43.32 | Get_The_Fish | ruben23 what type of connection tech are you on? |
| 22:43.58 | nargon | well its to late now i use it for our office server.. i use pure asterisk for our production stuff for our customers but I don't want to reprogram the 120 some odd extensions for our office users :( |
| 22:44.08 | nargon | it worked before untill they switched me to a business account |
| 22:45.28 | nargon | aside from the stupid web interface its astrisk underneath so.. it just has some stupid stuff going on with the "UI" right.. |
| 22:46.24 | ruben23 | Get_The_Fish:what you mean..? |
| 22:46.50 | Get_The_Fish | ruben23: t1, fiber, cable modem, etc |
| 22:46.52 | ruben23 | nargon: the host is the voip provider switch..? |
| 22:46.54 | nargon | i blame BV rather than "sh1tb0x" .. |
| 22:47.10 | ruben23 | Get_The_Fish:im using coppe |
| 22:47.15 | ruben23 | copper i mena |
| 22:47.25 | nargon | ruben23 i think your voip provider should not have latency issues i'm talking about your box on the 10Mbps link |
| 22:47.49 | Get_The_Fish | ruben23: ok, copper what? whats the transport? |
| 22:48.04 | nargon | you could start by making your net connection is solid by pinging your trix from somwhere on the net.. |
| 22:48.13 | nargon | making sure* |
| 22:48.55 | nargon | s/trix/asterisk box/ |
| 22:49.03 | ruben23 | Get_The_Fish: copper 10/100 Mbps |
| 22:51.14 | Get_The_Fish | so it's ethernet? Who is the carrier? |
| 22:52.34 | Get_The_Fish | Different technologies have different characteristics with regards to WAN connectivity. Some technologies are most burst than others, etc... a data T1 may not have much bandwidth but the underlying technology makes for a relatively jitter free connection. |
| 22:53.08 | ruben23 | nargon: --->http://pastebin.com/HTWukkKr |
| 22:53.54 | ruben23 | Get_The_Fish:yes ethernet, provider of the line bandwidth or the voip..? |
| 22:54.44 | Get_The_Fish | ruben23: provider of the line. The issue that you describe is related to jitter, or variations in the time between packets arriving at an endpoint. |
| 22:55.18 | Get_The_Fish | A connection that has a solid 80ms response will generally sound better than one that oscillates between 20 and 80 rapidly. |
| 22:56.51 | ruben23 | <PROTECTED> |
| 22:56.55 | nargon | ruben23 the 74. ip is your ast box and you pinged this from where ? |
| 22:57.17 | ruben23 | what causes the jitter..? is it my data traffic..? |
| 22:57.46 | ruben23 | nargon: i ping the voip carrier switch on US |
| 22:57.55 | nargon | 3.735 ms is your stdev/mdev so thats not much what codec are you using ? |
| 22:57.55 | ruben23 | im using my asterisk box |
| 22:58.24 | nargon | i suppose that would work f9.. you don't have much jitter really |
| 22:58.48 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-143-123.dsl.stlsmo.sbcglobal.net) |
| 22:58.51 | nargon | maybee your ast server hardware is weak how many calls are you processing at the same time ? |
| 22:59.02 | LemensTS | anyone know where i can get the lastest PAP2T firmware? |
| 22:59.08 | Get_The_Fish | nargon, out of curiosity where did you get that stdev number? |
| 22:59.35 | nargon | Get_The_Fish got it from his pastebin at the bottom of the ping output |
| 22:59.39 | *** part/#asterisk Scorcerer (scor@czlug.icis.pcz.pl) |
| 22:59.42 | ruben23 | nargon: how about this------> http://pastebin.com/XhzkB4cj |
| 23:00.20 | Get_The_Fish | oh he sent me a different pastebin is why... :) just checking |
| 23:01.05 | Get_The_Fish | ruben23, there are some hops there that are a little shaky but nothing too bad... |
| 23:01.17 | ruben23 | nargon: worst ping is 600 plus ms. |
| 23:01.18 | Get_The_Fish | ruben23 who is the itsp? |
| 23:01.26 | nargon | i added minexpirey, maxexpirey, defaultexpirey = 3600 to my peer definition but its still trying to register with a 120 s timeout |
| 23:01.55 | ruben23 | Get_The_Fish:im in the philippine sim using bayantel |
| 23:02.52 | Get_The_Fish | ruben23, mtr can be a little deceiving, as it is pinging each hope on the route, and most routers are equiped to prioritize traffic... obviously ICMP gets a lower priority |
| 23:03.02 | *** part/#asterisk devmod_ (~devmod__@c-76-100-208-204.hsd1.md.comcast.net) |
| 23:03.15 | nargon | 125.212.40.1 ruben23 is this your box ? |
| 23:03.17 | Get_The_Fish | are you doing conferencing at all? What is your timing source? |
| 23:03.35 | nargon | ruben23 give me your box ip i'll ping it from my nms in germany |
| 23:03.52 | Get_The_Fish | I can do the same here from the US |
| 23:05.41 | riddlebox | haha been fighting a phone not registering for an hour then I finally figured I should check iptables and the ip is blocked lol |
| 23:05.57 | *** join/#asterisk jks (jks@193.189.93.254) |
| 23:06.01 | nargon | nice.. |
| 23:06.07 | ruben23 | nargon: my box IP is 125.212.40.6 |
| 23:06.42 | ruben23 | Get_The_Fish: doing conference call -yes and my timing osurce is ztdummy. |
| 23:07.09 | Get_The_Fish | that can potentially be an issue, depending on the accuracy of the system clock |
| 23:07.12 | *** join/#asterisk knot (yiffstar66@unaffiliated/devemo) |
| 23:07.32 | nargon | round-trip min/avg/max/stddev = 390.664/404.720/416.555/5.655 ms with 2% loss.. from germany... |
| 23:07.35 | Get_The_Fish | ztdummy or dahdi dummy? What version of zaptel/dahdi/asterisk |
| 23:08.26 | nargon | looks horrible compaired to the ping between my germany and US NMSs |
| 23:09.06 | nargon | which is round-trip min/avg/max/stddev = 137.457/137.510/137.772/0.080 ms with 0% loss |
| 23:09.14 | Get_The_Fish | yeah mine gets pretty rough right before it hits the pacific |
| 23:10.07 | nargon | what codec are you using and will is sustain 2% loss with 6ms of jitter i guess is your question but i'm not the one to answer that.. |
| 23:10.56 | idespinner | 6ms jitter is fine |
| 23:11.08 | idespinner | usually below 20 is OK |
| 23:11.19 | idespinner | latency is ok until you hit around 50-100 |
| 23:11.30 | *** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no) |
| 23:12.06 | nargon | i run ulaw at 600ms latency on sat links with no real issue other than a slight delay 2% loss though might cause an issue... |
| 23:12.37 | idespinner | aslong as its stable and your buffer is large enough you should be ok |
| 23:12.50 | idespinner | but i'd be people notice delay when talking |
| 23:12.57 | idespinner | s/be/bet/ |
| 23:13.00 | ruben23 | <PROTECTED> |
| 23:13.05 | nargon | whats your proccessor look like ? |
| 23:13.23 | ruben23 | im uisng xeon dual core 2.4 Ghz, 4Gb ram |
| 23:13.26 | *** join/#asterisk x303 (~X303@28.28.102.97.cfl.res.rr.com) |
| 23:13.35 | nargon | ruben23 i mean what is the utilization |
| 23:13.47 | nargon | try doing an mpstat -P ALL to get a realtime readout |
| 23:14.56 | nargon | ruben23 try ulaw you wil use more bandwidth i think but you might bet better call quality ( i'm know very little off hand about codecs and there tolerance to loss) |
| 23:16.08 | nargon | idespinner yeah i notice the delay.. there is just enough delay to interupt people when talking.. |
| 23:18.09 | Get_The_Fish | ruben23, the absolute best way to troubleshoot this IMHO is to use tcpdump on the asterisk box, then analyze the results with wireshark... it will give you a clearer picture as to what is going on when you have issues. |
| 23:22.39 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 23:23.52 | nargon | can anyone tell me how to set the registration expiry for my peer, when i debug i see the sip registration attempts have expirty set to 120 even though i specified min max defaultexpiry = 3600 |
| 23:30.28 | *** join/#asterisk xuser (~xuser@unaffiliated/xuser) |
| 23:31.03 | nargon | also tried registertimeout=3600 not working |
| 23:34.23 | *** join/#asterisk glwgoes (~guilherme@189.114.202.44.dynamic.adsl.gvt.net.br) |
| 23:38.22 | nargon | defaultexpiry = 600 |
| 23:38.41 | nargon | can be set in global under sip.conf but that probably sets it for all peers.. |
| 23:39.01 | nargon | it workes though |
| 23:45.13 | *** part/#asterisk LemensTS (~LemensTS@adsl-70-238-143-123.dsl.stlsmo.sbcglobal.net) |
| 23:46.07 | [TK]D-Fender | nargon: Of course it has to be under [general], this is not a peer option |
| 23:46.25 | [TK]D-Fender | nargon: PEERS have nothing to do with registering |
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| 23:58.33 | saisoma | hey guys, quick question. I'm getting complaints that the polycom 331s are picking up too much background on my * 1.6.2.7 server. anyone know a setting off the top of their head for adjustment? |
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